From steveayre at gmail.com  Wed Jun  1 00:45:55 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 31 May 2011 21:45:55 +0100
Subject: [Freeswitch-users] how to use mod_mp4
In-Reply-To: <1306858803865-6423457.post@n2.nabble.com>
References: <1306486049569-6410226.post@n2.nabble.com>
	<BANLkTi=4ky6Kj9gE94k_i7tDvdW+LcpjvQ@mail.gmail.com>
	<1306841276439-6422324.post@n2.nabble.com>
	<BANLkTi=O5A_CfCO6PVeyz+RGOCA6Zv1nug@mail.gmail.com>
	<1306858803865-6423457.post@n2.nabble.com>
Message-ID: <BANLkTinUkgF1semv1VF1edY0WGBsKhJ67Q@mail.gmail.com>

I only heard of the module today, but let's take a look...

src/mod/applications/mod_mp4/Makefile contains:
LOCAL_LDFLAGS=-lmp4v2

So it's trying to use the mp4v2 library, probably packaged as libmp4v2-dev
or libmp4v2-devel, or something similar

It's website is http://resare.com/libmp4v2/ if your distribution doesn't
have a package.

-Steve




On 31 May 2011 17:20, mazilo <Nabble at slickdeals.endjunk.com> wrote:

> Steven,
>
> Do you know what library package is needed to compile applications/mod_mp4?
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3
> Watts of electricity.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/how-to-use-mod-mp4-tp6410226p6423457.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Wed Jun  1 00:58:26 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 31 May 2011 13:58:26 -0700
Subject: [Freeswitch-users] Any way to limit the ability to create
 conference rooms on demand?
In-Reply-To: <4DE3532A.9000004@omigos.de>
References: <4DE3532A.9000004@omigos.de>
Message-ID: <BANLkTikMwQePifbiVMPdcSg9nnEKUdG_Uw@mail.gmail.com>

There are only two practical ways to do this: hard-code your conference
rooms in the dialplan or use a database-backed solution like mod_xml_curl.
How are you currently managing the list of available conferences?

-MC

2011/5/30 Andr? Rosowski <andre.rosowski at omigos.de>

> Hi there,
>
> I'm using Bigbluebutton (online conferencing) with Freeswitch so that
> users can call a number and then have to enter the "voicebridge" number
> to be put into a voice conference. The problem, however, is that if a
> user dialed the wrong "voicebridge" number he will be put into a new
> conference thus creating one on demand as stated in the describtion of
> "mod_conference". Is there any way to manually create conference rooms
> and make them static...not allowing for new conference "rooms" to be
> created on demand.
>
> default.xml in dialplan:
>
> <extension name="phone conference">
> <condition field="destination_number" expression="^(SEND_TO_CONFERENCE)$">
> <action application="conference" data="${pin}@wideband"/>
> </condition>
> </extension>
>
> Thanks for your help : )
>
> Regards,
>
> Realdoe
>
> --
> Omigos Labs UG (haftungsbeschr?nkt) - Meisenstra?e 96 - 33607 Bielefeld
> Fon: +49 521 2997 200 - Fax: +49 521 2997 101
> info at omigos.de - www.omigos.de
> Gesch?ftsf?hrer: BSc Andr? Rosowski, Benjamin Bittner, Julian Klima
> Registergericht: Amtsgericht Bielefeld - Registernummer: HRB 40255
> USt-IdNr.: 305/5860/1585
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Wed Jun  1 01:10:58 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 31 May 2011 14:10:58 -0700
Subject: [Freeswitch-users] Bridging Gateway from another server
In-Reply-To: <4DE36FB2.5080802@xofap.com>
References: <4DDE2633.2030002@xofap.com>
	<BANLkTinGsRnpx_g6V3-Lk1BZT2XwayvRXg@mail.gmail.com>
	<4DDF6FFA.9060108@xofap.com>
	<BANLkTi=XnmTVO-DWqAdcCMY-gWCvrOZCOw@mail.gmail.com>
	<4DE36FB2.5080802@xofap.com>
Message-ID: <BANLkTiktiWAz0GqtsuhNVWQN+NBWZFMsVw@mail.gmail.com>

We still need to see a debug w/ sip trace if you want us to help you with
this.
-MC

On Mon, May 30, 2011 at 3:21 AM, William Alianto <william at xofap.com> wrote:

>  I found out that it's codec problem. I enabled the wrong codec, so it
> didn't connect. After I fixed the codec, the call still not successful. I
> didn't see any incoming log on FS2 either.
>
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From freeswitch-list at puzzled.xs4all.nl  Wed Jun  1 02:06:43 2011
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Wed, 01 Jun 2011 00:06:43 +0200
Subject: [Freeswitch-users] Freeswitch tts russian voice 'ele' or other?
In-Reply-To: <BANLkTik7ZHmvCXu=3fH3=8AdjE29ic9nJg@mail.gmail.com>
References: <BANLkTik7ZHmvCXu=3fH3=8AdjE29ic9nJg@mail.gmail.com>
Message-ID: <4DE56673.4090109@puzzled.xs4all.nl>

On 05/31/2011 08:12 PM, Anton VG wrote:
> Hi!
>
> In ru.xml it's mentioned
>
> tts-engine="cepstral" tts-voice="elena">
>
> But after hours of googling I found nothing where and how to get the
> mentioned russian 'elena' voice for cepstral
>
> Any info?

There are Elena sound files at http://files.freeswitch.org/

Regards,
Patrick



From marketing at cluecon.com  Wed Jun  1 02:08:55 2011
From: marketing at cluecon.com (marketing at cluecon.com)
Date: Tue, 31 May 2011 22:08:55 +0000
Subject: [Freeswitch-users] ClueCon 2011 News and Notes
Message-ID: <00000130481a3868-69cf136f-c857-407d-90c0-e75120e27c52-000000@email.amazonses.com>

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From anton.vazir at gmail.com  Wed Jun  1 08:32:59 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Wed, 1 Jun 2011 09:32:59 +0500
Subject: [Freeswitch-users] Freeswitch tts russian voice 'ele' or other?
In-Reply-To: <4DE56673.4090109@puzzled.xs4all.nl>
References: <BANLkTik7ZHmvCXu=3fH3=8AdjE29ic9nJg@mail.gmail.com>
	<4DE56673.4090109@puzzled.xs4all.nl>
Message-ID: <BANLkTim9yzBLFDpO-yKbS0SKvUQOFrfmTQ@mail.gmail.com>

This is sounds, I'm looking for TTS voice

2011/6/1 Patrick Lists <freeswitch-list at puzzled.xs4all.nl>:
> On 05/31/2011 08:12 PM, Anton VG wrote:
>> Hi!
>>
>> In ru.xml it's mentioned
>>
>> tts-engine="cepstral" tts-voice="elena">
>>
>> But after hours of googling I found nothing where and how to get the
>> mentioned russian 'elena' voice for cepstral
>>
>> Any info?
>
> There are Elena sound files at http://files.freeswitch.org/
>
> Regards,
> Patrick
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From zetruger at gmail.com  Wed Jun  1 10:19:05 2011
From: zetruger at gmail.com (=?KOI8-R?B?6dfBziD+ydPU0cvP1w==?=)
Date: Wed, 1 Jun 2011 10:19:05 +0400
Subject: [Freeswitch-users] Freeswitch tts russian voice 'ele' or other?
In-Reply-To: <BANLkTim9yzBLFDpO-yKbS0SKvUQOFrfmTQ@mail.gmail.com>
References: <BANLkTik7ZHmvCXu=3fH3=8AdjE29ic9nJg@mail.gmail.com>
	<4DE56673.4090109@puzzled.xs4all.nl>
	<BANLkTim9yzBLFDpO-yKbS0SKvUQOFrfmTQ@mail.gmail.com>
Message-ID: <BANLkTi=Qqd5YXdcpE-ZrNq8+AU3V6KVT1w@mail.gmail.com>

I am using festival.


sudo apt-get install festival festvox-ru

<configuration name="tts_commandline.conf" description="TextToSpeech
Commandline configuration">
    <settings>
	<param name="command" value="echo ${text} | text2wave -eval
'(${voice})' -f ${rate} > ${file}"/>
    </settings>
</configuration>

session.execute("speak", "tts_commandline|voice_msu_ru_nsh_clunits|??????.")



2011/6/1 Anton VG <anton.vazir at gmail.com>:
> This is sounds, I'm looking for TTS voice
>
> 2011/6/1 Patrick Lists <freeswitch-list at puzzled.xs4all.nl>:
>> On 05/31/2011 08:12 PM, Anton VG wrote:
>>> Hi!
>>>
>>> In ru.xml it's mentioned
>>>
>>> tts-engine="cepstral" tts-voice="elena">
>>>
>>> But after hours of googling I found nothing where and how to get the
>>> mentioned russian 'elena' voice for cepstral
>>>
>>> Any info?
>>
>> There are Elena sound files at http://files.freeswitch.org/
>>
>> Regards,
>> Patrick
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From infos at madovsky.org  Wed Jun  1 10:23:24 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 1 Jun 2011 02:23:24 -0400
Subject: [Freeswitch-users] mod_msn
Message-ID: <931F80EC287F4BBEA4174FFBE344D831@e1705>

Do you think it's technically feasible to
make a module that connect to MSN network ?
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From sid.kshatriya at gmail.com  Wed Jun  1 10:47:33 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Wed, 1 Jun 2011 12:17:33 +0530
Subject: [Freeswitch-users] mod_msn
In-Reply-To: <931F80EC287F4BBEA4174FFBE344D831@e1705>
References: <931F80EC287F4BBEA4174FFBE344D831@e1705>
Message-ID: <BANLkTi=QsUA571L9bq6sSesG2FMtV+20KA@mail.gmail.com>

What will the module do? And why not -- as long as you are able to write C
code for it... its possible...

On Wed, Jun 1, 2011 at 11:53 AM, Madovsky <infos at madovsky.org> wrote:

>  Do you think it's technically feasible to
> make a module that connect to MSN network ?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sidharth Kshatriya
www.sidk.info
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From curriegrad2004 at gmail.com  Wed Jun  1 10:48:35 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Tue, 31 May 2011 23:48:35 -0700
Subject: [Freeswitch-users] mod_msn
In-Reply-To: <931F80EC287F4BBEA4174FFBE344D831@e1705>
References: <931F80EC287F4BBEA4174FFBE344D831@e1705>
Message-ID: <BANLkTim79fhJqAYv=71-hzwc5Prpm8twiw@mail.gmail.com>

When I first read that line I thought, oh, my anthm is at it again...

Technically I think is feasible for such a module. We could use
libpurple to interface it with FS, but only the text part has been
done so far.

On Tue, May 31, 2011 at 11:23 PM, Madovsky <infos at madovsky.org> wrote:
> Do you think it's technically feasible to
> make a module that connect to MSN network ?
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From anton.vazir at gmail.com  Wed Jun  1 10:53:09 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Wed, 1 Jun 2011 11:53:09 +0500
Subject: [Freeswitch-users] Freeswitch tts russian voice 'ele' or other?
In-Reply-To: <BANLkTi=Qqd5YXdcpE-ZrNq8+AU3V6KVT1w@mail.gmail.com>
References: <BANLkTik7ZHmvCXu=3fH3=8AdjE29ic9nJg@mail.gmail.com>
	<4DE56673.4090109@puzzled.xs4all.nl>
	<BANLkTim9yzBLFDpO-yKbS0SKvUQOFrfmTQ@mail.gmail.com>
	<BANLkTi=Qqd5YXdcpE-ZrNq8+AU3V6KVT1w@mail.gmail.com>
Message-ID: <BANLkTi=XuYcjHm94bJKvOQe2ZV09+6J_gw@mail.gmail.com>

Ivan, The question is specific on 'elena' or other russian, business
quality voice for cepstral, since it confusingly mentioned in tts
settings.

1 ???? 2011??. 11:19 ???????????? ???? ???????? <zetruger at gmail.com> ???????:
> I am using festival.
>
>
> sudo apt-get install festival festvox-ru
>
> <configuration name="tts_commandline.conf" description="TextToSpeech
> Commandline configuration">
> ? ?<settings>
> ? ? ? ?<param name="command" value="echo ${text} | text2wave -eval
> '(${voice})' -f ${rate} > ${file}"/>
> ? ?</settings>
> </configuration>
>
> session.execute("speak", "tts_commandline|voice_msu_ru_nsh_clunits|??????.")
>
>
>
> 2011/6/1 Anton VG <anton.vazir at gmail.com>:
>> This is sounds, I'm looking for TTS voice
>>
>> 2011/6/1 Patrick Lists <freeswitch-list at puzzled.xs4all.nl>:
>>> On 05/31/2011 08:12 PM, Anton VG wrote:
>>>> Hi!
>>>>
>>>> In ru.xml it's mentioned
>>>>
>>>> tts-engine="cepstral" tts-voice="elena">
>>>>
>>>> But after hours of googling I found nothing where and how to get the
>>>> mentioned russian 'elena' voice for cepstral
>>>>
>>>> Any info?
>>>
>>> There are Elena sound files at http://files.freeswitch.org/
>>>
>>> Regards,
>>> Patrick
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From infos at madovsky.org  Wed Jun  1 10:59:06 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 1 Jun 2011 02:59:06 -0400
Subject: [Freeswitch-users] mod_msn
References: <931F80EC287F4BBEA4174FFBE344D831@e1705>
	<BANLkTi=QsUA571L9bq6sSesG2FMtV+20KA@mail.gmail.com>
Message-ID: <26661ECBCB7E455A8F293DBA50FC4D8A@e1705>

text and audio/video
  ----- Original Message ----- 
  From: Sidharth Kshatriya 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, June 01, 2011 2:47 AM
  Subject: Re: [Freeswitch-users] mod_msn


  What will the module do? And why not -- as long as you are able to write C code for it... its possible...


  On Wed, Jun 1, 2011 at 11:53 AM, Madovsky <infos at madovsky.org> wrote:

    Do you think it's technically feasible to
    make a module that connect to MSN network ?

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





  -- 
  Sidharth Kshatriya
  www.sidk.info




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


  _______________________________________________
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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From awais-nazeer at hotmail.com  Wed Jun  1 12:33:53 2011
From: awais-nazeer at hotmail.com (awais nazir)
Date: Wed, 1 Jun 2011 14:33:53 +0600
Subject: [Freeswitch-users] Call jumping
Message-ID: <BAY156-w14FAC14F0F4B2C97CE942F887D0@phx.gbl>


Hi

I want to achieve a scenario like 

Calls try to hit on first carrier and if return code is 503 or 504 (may be more) then it should hit on second carrier (carriers list can also go on)
 		 	   		  
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From peter.olsson at visionutveckling.se  Wed Jun  1 12:52:46 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 1 Jun 2011 10:52:46 +0200
Subject: [Freeswitch-users] Call jumping
In-Reply-To: <BAY156-w14FAC14F0F4B2C97CE942F887D0@phx.gbl>
References: <BAY156-w14FAC14F0F4B2C97CE942F887D0@phx.gbl>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59DE5494D3@cooper>

Check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge

It basically works like this; <action application="bridge" data="sofia/gateway/gw1/$1| sofia/gateway/gw2/$1"/>

/Peter

Mvh
Peter Olsson

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r awais nazir
Skickat: den 1 juni 2011 10:34
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] Call jumping

Hi

I want to achieve a scenario like

Calls try to hit on first carrier and if return code is 503 or 504 (may be more) then it should hit on second carrier (carriers list can also go on)
!DSPAM:4de5fa9e32761556550103!
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From peter.olsson at visionutveckling.se  Wed Jun  1 13:06:41 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 1 Jun 2011 11:06:41 +0200
Subject: [Freeswitch-users] ClueCon 2011 News and Notes
In-Reply-To: <00000130481a3868-69cf136f-c857-407d-90c0-e75120e27c52-000000@email.amazonses.com>
References: <00000130481a3868-69cf136f-c857-407d-90c0-e75120e27c52-000000@email.amazonses.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59DE5494FE@cooper>

I booked everything today, for me and a collegue - I'm really looking forward to it!

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r marketing at cluecon.com
Skickat: den 1 juni 2011 00:09
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] ClueCon 2011 News and Notes

ClueCon 2011 Reminder: Extra Chances to Win

We wanted to remind everyone that you still have one more day to book ClueCon 2011 and receive FOUR entries into big giveaways for the coolest laptop and iPad<http://www.cluecon.com/coolest>. (Promotion ends on June 1.) Will you be the big winner? Book now and increase your chances!

For those wanting to know more about what's happening at ClueCon this year we've posted our preliminary schedule<http://www.cluecon.com/schedule>. There are some great talks planned and, of course, a few surprises. Be sure to check back often to see the latest speakers and sponsors.

Visit us at http://www.cluecon.com for more information, or call us at 877.742.CLUE (2583) to inquire about registration, sponsorships, or any other questions you may have. See you in August!

The ClueCon Team
http://www.cluecon.com
877.742.CLUE

[cid:~WRD000.jpg]<http://www.cluecon.com>

!DSPAM:4de567b932761932211873!
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From chrisg.lists at gmail.com  Wed Jun  1 13:45:45 2011
From: chrisg.lists at gmail.com (Chris Graham)
Date: Wed, 1 Jun 2011 11:45:45 +0200
Subject: [Freeswitch-users] XML error with static and dynamic XML
Message-ID: <BANLkTim-BLo4Mv1oCJoBeocS2NhkgQ7GsA@mail.gmail.com>

Hi List,

Apologies if this is received twice..

I am getting the malformed XML error below:

2011-05-31 13:32:09.392074 [INFO] mod_dialplan_xml.c:331 Processing
1000 <1000>->0112341111 in context lcr_trunks
2011-05-31 13:32:09.432080 [ERR] switch_xml.c:1611 Error[[error near
line 11]: unexpected closing tag </document>]
2011-05-31 13:32:09.432080 [WARNING] mod_dialplan_xml.c:361 Context
lcr_trunks not found
2011-05-31 13:32:09.432080 [INFO] switch_core_state_machine.c:142 No
Route, Aborting


My static XML file being called:
<?xml version="1.0" encoding="utf-8"?>
<include>
       <context name="default" />
</include>

My dynamic XML being served up by xml_curl:
<?xml version="1.0" encoding="UTF-8" standalone="no"?><document
type="freeswitch/xml">
 <section name="dialplan" description="RE Dial Plan For FreeSwitch">
   <context name="lcr_trunks">
     <extension name="Catch carrier test">
       <condition field="destination_number" expression="^(.*)$">
         <action application="lcr" data="$1 peak"/>
         <action application="bridge" data="${lcr_auto_route}"/>
       </condition>
     </extension>
   </context>
 </section>
</document>

Why is the dynamic XML having an issue with "</document>" tag? Looks
legal to me?

Thanks in advance,
Chris


From steveayre at gmail.com  Wed Jun  1 13:57:13 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 1 Jun 2011 10:57:13 +0100
Subject: [Freeswitch-users] mod_msn
In-Reply-To: <26661ECBCB7E455A8F293DBA50FC4D8A@e1705>
References: <931F80EC287F4BBEA4174FFBE344D831@e1705>
	<BANLkTi=QsUA571L9bq6sSesG2FMtV+20KA@mail.gmail.com>
	<26661ECBCB7E455A8F293DBA50FC4D8A@e1705>
Message-ID: <BANLkTi=ye7biz3c8qLdQUW-1Lei3LmkaaQ@mail.gmail.com>

Text should be simple the protocol (
http://en.wikipedia.org/wiki/Microsoft_Notification_Protocol) has been
successfully reverse engineered and used by several open source projects
(e.g. Pidgin, libmsn).

Audio/video is probably not possible because that hasn't been reverse
engineered.

Sounds like Microsoft are working on developing interop with Google Talk
though, so mod_dingaling might be able to be used in future (but not yet).
http://www.liveside.net/2007/10/31/messenger-9-gtalk-integration-messenger-api-new-client-for-mac-os-x-news-unveiled-at-georgia-tech-presentation-whew/

-Steve



On 1 June 2011 07:59, Madovsky <infos at madovsky.org> wrote:

>  text and audio/video
>
> ----- Original Message -----
> *From:* Sidharth Kshatriya <sid.kshatriya at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, June 01, 2011 2:47 AM
> *Subject:* Re: [Freeswitch-users] mod_msn
>
> What will the module do? And why not -- as long as you are able to write C
> code for it... its possible...
>
> On Wed, Jun 1, 2011 at 11:53 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  Do you think it's technically feasible to
>> make a module that connect to MSN network ?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Sidharth Kshatriya
> www.sidk.info
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Nabble at slickdeals.endjunk.com  Wed Jun  1 15:17:28 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 1 Jun 2011 04:17:28 -0700 (PDT)
Subject: [Freeswitch-users] how to use mod_mp4
In-Reply-To: <BANLkTinUkgF1semv1VF1edY0WGBsKhJ67Q@mail.gmail.com>
References: <1306486049569-6410226.post@n2.nabble.com>
	<BANLkTi=4ky6Kj9gE94k_i7tDvdW+LcpjvQ@mail.gmail.com>
	<1306841276439-6422324.post@n2.nabble.com>
	<BANLkTi=O5A_CfCO6PVeyz+RGOCA6Zv1nug@mail.gmail.com>
	<1306858803865-6423457.post@n2.nabble.com>
	<BANLkTinUkgF1semv1VF1edY0WGBsKhJ67Q@mail.gmail.com>
Message-ID: <1306927048156-6426476.post@n2.nabble.com>


Steven Ayre wrote:
> 
> I only heard of the module today, but let's take a look...
> 
> src/mod/applications/mod_mp4/Makefile contains:
> LOCAL_LDFLAGS=-lmp4v2
> 
> So it's trying to use the mp4v2 library, probably packaged as libmp4v2-dev
> or libmp4v2-devel, or something similar
> 
> It's website is http://resare.com/libmp4v2/ if your distribution doesn't
> have a package.
Steve, thank you.

Apparently, the development of libmp4v2 stopped @ v1.6.1 back in 2007.
Hopefully, the new  http://code.google.com/p/mp4v2 mp4v2  will work just
fine.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/how-to-use-mod-mp4-tp6410226p6426476.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Wed Jun  1 15:36:11 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 1 Jun 2011 12:36:11 +0100
Subject: [Freeswitch-users] how to use mod_mp4
In-Reply-To: <1306927048156-6426476.post@n2.nabble.com>
References: <1306486049569-6410226.post@n2.nabble.com>
	<BANLkTi=4ky6Kj9gE94k_i7tDvdW+LcpjvQ@mail.gmail.com>
	<1306841276439-6422324.post@n2.nabble.com>
	<BANLkTi=O5A_CfCO6PVeyz+RGOCA6Zv1nug@mail.gmail.com>
	<1306858803865-6423457.post@n2.nabble.com>
	<BANLkTinUkgF1semv1VF1edY0WGBsKhJ67Q@mail.gmail.com>
	<1306927048156-6426476.post@n2.nabble.com>
Message-ID: <BANLkTimLwBeTnNi13zNhZnH9KU4uLWTPwA@mail.gmail.com>

Should do, it's the same library. :)

Much better website you found there - I just pulled the first one I saw out
of Google.

-Steve


On 1 June 2011 12:17, mazilo <Nabble at slickdeals.endjunk.com> wrote:

>
> Steven Ayre wrote:
> >
> > I only heard of the module today, but let's take a look...
> >
> > src/mod/applications/mod_mp4/Makefile contains:
> > LOCAL_LDFLAGS=-lmp4v2
> >
> > So it's trying to use the mp4v2 library, probably packaged as
> libmp4v2-dev
> > or libmp4v2-devel, or something similar
> >
> > It's website is http://resare.com/libmp4v2/ if your distribution doesn't
> > have a package.
> Steve, thank you.
>
> Apparently, the development of libmp4v2 stopped @ v1.6.1 back in 2007.
> Hopefully, the new  http://code.google.com/p/mp4v2 mp4v2  will work just
> fine.
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3
> Watts of electricity.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/how-to-use-mod-mp4-tp6410226p6426476.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From freeswitch at simpot.com  Wed Jun  1 17:45:41 2011
From: freeswitch at simpot.com (Dmitry Saratsky)
Date: Wed, 1 Jun 2011 16:45:41 +0300
Subject: [Freeswitch-users] FS vs provider's Sonus DTMF (double digit)
	problem
Message-ID: <000001cc2062$3ae1c460$b0a54d20$@com>

Hi All,

 

I'm suffering from dtmf "double digit" problem for incoming calls that
destined to IVR on my FS.

I tryed to identify and workaround the provlem with no success...

I have already tryed to change dtmf-mode, use sonus-fixup etc with no
luck...

I have: FreeSWITCH Version 1.0.head (git-1086cba 2011-05-23 22-51-43 -0500)

I would like to ask you some questions and I'm attaching some relevant
output (see below)

 

1. According to output (see below), is it right to say, that my provider
sends me DTMF events both in "DTMF INFO" and "RTP DTMF - rfc2833" way?

2. If so, may be this is the reason for "double digit" I suffer?

3. If so, can I filter incoming "DTMF INFO" events in FS?

4. If so, do FS have some workaround for this?

 

Any ideas?

 

Thanks,

Dmitry.

 

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

 

FreeSWITCH Version 1.0.head (git-1086cba 2011-05-23 22-51-43 -0500)

-------

 

<profile name="providers">
  <gateways>
    <X-PRE-PROCESS cmd="include" data="providers/*.xml"/>
  </gateways>

  <aliases>
  </aliases>

  <domains>
    <domain name="all" alias="false" parse="true"/>
  </domains>

  <settings>
    <param name="debug" value="0"/>
    <param name="shutdown-on-fail" value="true"/>
    <param name="auth-calls" value="false"/>
    <param name="sip-trace" value="no"/>
    <param name="disable-register" value="true"/>
    <param name="rfc2833-pt" value="101"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="providers"/>
    <param name="apply-inbound-acl" value="013SONUS-IPS"/>
    <param name="hold-music" value="$${hold_music}"/>
    <param name="stun-enabled" value="false"/>
    <param name="stun-auto-disable" value="true"/>
    <param name="manage-presence" value="false"/>
    <param name="nonce-ttl" value="60"/>
    <param name="dtmf-type" value="rfc2833"/>
    <param name="dtmf-duration" value="2000"/>

    <param name="rtp-timer-name" value="soft"/>
    <param name="rtp-timeout-sec" value="50"/>
    <param name="rtp-hold-timeout-sec" value="300"/>

    <param name="sip-port" value="$${sip_port}"/>

    <param name="rtp-ip" value="$${rtp_ip}"/>
    <param name="sip-ip" value="$${ext_ip}"/>

    <param name="inbound-codec-prefs" value="PCMU,PCMA"/>
    <param name="outbound-codec-prefs" value="PCMU,PCMA"/>
    <param name="codec-prefs" value="PCMU,PCMA"/>
    <param name="inbound-late-negotiation" value="false"/>
    <param name="inbound-codec-negotiation" value="scrooge"/>
    <param name="disable-transcoding" value="false"/>

  </settings>
</profile>

-------

 

2011-06-01 10:21:30.585146 [DEBUG] sofia.c:6236 INFO DTMF(0)
2011-06-01 10:21:30.585146 [DEBUG] sofia.c:6310 dispatched freeswitch event
for INFO
2011-06-01 10:21:30.604196 [DEBUG] switch_ivr_play_say.c:1649 done playing
file
2011-06-01 10:21:30.604196 [DEBUG] switch_ivr_menu.c:343 waiting for 2/3
digits t/o 2000
2011-06-01 10:21:30.845091 [DEBUG] sofia.c:6207 Bad signal
2011-06-01 10:21:30.845091 [DEBUG] sofia.c:6310 dispatched freeswitch event
for INFO
2011-06-01 10:21:30.966061 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF 0:4360
2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:390 digits '00'
2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:484 action regex [00]
[/^(20[1-2])$/] [0]
2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:484 action regex [00]
[/^[3-9*#]$/] [0]
2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:574 IVR menu
'main_ivr_heb' caught invalid input '00'

 

-------


2011-06-01 10:23:23.902331 [DEBUG] sofia.c:6236 INFO DTMF(7)
2011-06-01 10:23:23.902331 [DEBUG] sofia.c:6310 dispatched freeswitch event
for INFO
2011-06-01 10:23:23.922329 [DEBUG] switch_ivr_play_say.c:1649 done playing
file
2011-06-01 10:23:23.922329 [DEBUG] switch_ivr_menu.c:343 waiting for 2/3
digits t/o 2000
2011-06-01 10:23:23.942304 [DEBUG] sofia.c:6207 Bad signal
2011-06-01 10:23:23.942304 [DEBUG] sofia.c:6310 dispatched freeswitch event
for INFO
2011-06-01 10:23:24.083260 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF 7:2760
2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:390 digits '77'
2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:484 action regex [77]
[/^(20[1-2])$/] [0]
2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:484 action regex [77]
[/^[3-9*#]$/] [0]
2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:574 IVR menu
'main_ivr_heb' caught invalid input '77'

 

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From vkozak at abisoft.spb.ru  Wed Jun  1 18:50:06 2011
From: vkozak at abisoft.spb.ru (Kozak Vladimir)
Date: Wed, 1 Jun 2011 18:50:06 +0400
Subject: [Freeswitch-users] PLAY_AND_GET_DIGITS dialplann application: set
	<lang> and <say_type> to SAY subcommand
Message-ID: <DF4F093EDA9F48BC9C688D75AB774225@abisoft.biz>

Hello all.
In example for "play_and_get_digits" dialplan app I see next extension. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits
<extension name="play_and_get_digits with say">
  <condition field="destination_number" expression="^(6500)$">
    <action application="play_and_get_digits" data="1 1 1 3000 # say:'press one for technicial support' silence_stream://250 res \d+" />
  </condition>
</extension>This extension say somethin and get user DTMF.
I need say digits and numbers on various (EN and ES) languages.
It's possible to specify these vars for "say" dialplan application http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_say. And it works correctly.

Could you tell me How can I set language and say_type for tts_engine in "play_and_get_digits" app case?

I try to use next extension, but it don't work (FS take 'es NUMBER 1234567'  as <text_to_say> and say all this text on English):
 <extension name="read_extension3">
        <condition field="destination_number" expression="^7890$">
            <action application="answer"/>
                <action application="set" data="tts_engine=flite"/>
                <action application="set" data="tts_voice=kal"/>
 <action application="play_and_get_digits" data="1 1 1 3000 # say:'es NUMBER 1234567' /usr/local/freeswitch/sounds/starpound/silence.wav read_result \d+" />
            <action application="event" data="Event-App-Type=READ-EXECUTED,Read-Result=${read_result}"/>
            <action application="log" data="Read-Result=${read_result}"/>
            <action application="park"/>
        </condition>
    </extension>

Also I try to set vars:
                <action application="set" data="tts_engine=cepstral"/>
                <action application="set" data="tts_voice=maria"/>        <!-- ES lang -->
and get failed result:
EXECUTE sofia/internal/1007 at 172.26.200.250:5060 play_and_get_digits(1 1 1 3000 # say:'1234567' /usr/local/freeswitch/sounds/starpound/silence.wav read_result \d+)
2011-06-05 22:58:33.661014 [ERR] switch_core_speech.c:61 Invalid speech module [cepstral]!
2011-06-05 22:58:33.661014 [ERR] switch_ivr_play_say.c:2361 Invalid TTS module!
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From msc at freeswitch.org  Wed Jun  1 19:14:34 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 1 Jun 2011 08:14:34 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <BANLkTimJKpFqEtBj42yE_721dKhUS2d+bg@mail.gmail.com>

Hello all!

Here is today's agenda page:

http://wiki.freeswitch.org/wiki/FS_weekly_2011_06_01

We have a guest from Shacast.com (Vuong Nguyen) who will be talking to us.
Visit http://www.shacast.com and click on the PowerPoint download link to
download the PPT that Vuong will be referencing for his presentation.

Thanks,
Michael
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From simpot at gmail.com  Wed Jun  1 11:40:24 2011
From: simpot at gmail.com (Dmitry Saratsky)
Date: Wed, 1 Jun 2011 10:40:24 +0300
Subject: [Freeswitch-users] FS vs provider's Sonus DTMF (double digit)
	problem
Message-ID: <BANLkTi=bwO5AUw4Py3W+4761_1w7TwFUGQ@mail.gmail.com>

Hi All,

I'm suffering from dtmf "double digit" problem for incoming calls that
destined to IVR on my FS.
I tryed to identify and workaround the provlem with no success...
I have already tryed to change dtmf-mode, use sonus-fixup etc with no
luck...
I have: FreeSWITCH Version 1.0.head (git-1086cba 2011-05-23 22-51-43 -0500)
I would like to ask you some questions and I'm attaching some relevant
output (see below)

1. According to output (see below), is it right to say, that my provider
sends me DTMF events both in "DTMF INFO" and "RTP DTMF - rfc2833" way?
2. If so, may be this is the reason for "double digit" I suffer?
3. If so, can I filter incoming "DTMF INFO" events in FS?
4. If so, do FS have some workaround for this?

Thanks,
Dmitry.

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

FreeSWITCH Version 1.0.head (git-1086cba 2011-05-23 22-51-43 -0500)
-------

<profile name="providers">
  <gateways>
    <X-PRE-PROCESS cmd="include" data="providers/*.xml"/>
  </gateways>
  <aliases>
  </aliases>
  <domains>
    <domain name="all" alias="false" parse="true"/>
  </domains>
  <settings>
    <param name="debug" value="0"/>
    <param name="shutdown-on-fail" value="true"/>
    <param name="auth-calls" value="false"/>
    <param name="sip-trace" value="no"/>
    <param name="disable-register" value="true"/>
    <param name="rfc2833-pt" value="101"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="providers"/>
    <param name="apply-inbound-acl" value="013SONUS-IPS"/>
    <param name="hold-music" value="$${hold_music}"/>
    <param name="stun-enabled" value="false"/>
    <param name="stun-auto-disable" value="true"/>
    <param name="manage-presence" value="false"/>
    <param name="nonce-ttl" value="60"/>
    <param name="dtmf-type" value="rfc2833"/>
    <param name="dtmf-duration" value="2000"/>
    <param name="rtp-timer-name" value="soft"/>
    <param name="rtp-timeout-sec" value="50"/>
    <param name="rtp-hold-timeout-sec" value="300"/>
    <param name="sip-port" value="$${sip_port}"/>
    <param name="rtp-ip" value="$${rtp_ip}"/>
    <param name="sip-ip" value="$${ext_ip}"/>
    <param name="inbound-codec-prefs" value="PCMU,PCMA"/>
    <param name="outbound-codec-prefs" value="PCMU,PCMA"/>
    <param name="codec-prefs" value="PCMU,PCMA"/>
    <param name="inbound-late-negotiation" value="false"/>
    <param name="inbound-codec-negotiation" value="scrooge"/>
    <param name="disable-transcoding" value="false"/>
  </settings>
</profile>
-------

2011-06-01 10:21:30.585146 [DEBUG] sofia.c:6236 INFO DTMF(0)
2011-06-01 10:21:30.585146 [DEBUG] sofia.c:6310 dispatched freeswitch event
for INFO
2011-06-01 10:21:30.604196 [DEBUG] switch_ivr_play_say.c:1649 done playing
file
2011-06-01 10:21:30.604196 [DEBUG] switch_ivr_menu.c:343 waiting for 2/3
digits t/o 2000
2011-06-01 10:21:30.845091 [DEBUG] sofia.c:6207 Bad signal
2011-06-01 10:21:30.845091 [DEBUG] sofia.c:6310 dispatched freeswitch event
for INFO
2011-06-01 10:21:30.966061 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF 0:4360
2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:390 digits '00'
2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:484 action regex [00]
[/^(20[1-2])$/] [0]
2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:484 action regex [00]
[/^[3-9*#]$/] [0]
2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:574 IVR menu
'main_ivr_heb' caught invalid input '00'

-------

2011-06-01 10:23:23.902331 [DEBUG] sofia.c:6236 INFO DTMF(7)
2011-06-01 10:23:23.902331 [DEBUG] sofia.c:6310 dispatched freeswitch event
for INFO
2011-06-01 10:23:23.922329 [DEBUG] switch_ivr_play_say.c:1649 done playing
file
2011-06-01 10:23:23.922329 [DEBUG] switch_ivr_menu.c:343 waiting for 2/3
digits t/o 2000
2011-06-01 10:23:23.942304 [DEBUG] sofia.c:6207 Bad signal
2011-06-01 10:23:23.942304 [DEBUG] sofia.c:6310 dispatched freeswitch event
for INFO
2011-06-01 10:23:24.083260 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF 7:2760
2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:390 digits '77'
2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:484 action regex [77]
[/^(20[1-2])$/] [0]
2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:484 action regex [77]
[/^[3-9*#]$/] [0]
2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:574 IVR menu
'main_ivr_heb' caught invalid input '77'
2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_play_say.c:1279 Codec
Activated L16 at 8000hz 1 channels 20ms
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From yungwei at resolvity.com  Wed Jun  1 20:52:32 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Wed, 1 Jun 2011 12:52:32 -0400
Subject: [Freeswitch-users] Transferring to a cell phone
In-Reply-To: <BANLkTi=4fMHUkHjoCZ8exuw6Qgg5Pgj6Cw@mail.gmail.com>
References: <33095823FD21DF429B481B5163264B7950AC3AC74C@VMBX102.ihostexchange.net>
	<0454EB31-E9EA-4594-9FDD-29D69F5B6126@ipeva.fr>
	<33095823FD21DF429B481B5163264B7950AC3AC7E2@VMBX102.ihostexchange.net>
	<1306499299750-6410887.post@n2.nabble.com>
	<33095823FD21DF429B481B5163264B7950AC3AC8B0@VMBX102.ihostexchange.net>
	<BANLkTikrgnaTDL9+WtXh_4RAWTFnmja71Q@mail.gmail.com>
	<33095823FD21DF429B481B5163264B7950AC3AC938@VMBX102.ihostexchange.net>
	<33095823FD21DF429B481B5163264B7950AC3ACA1B@VMBX102.ihostexchange.net>,
	<BANLkTi=4fMHUkHjoCZ8exuw6Qgg5Pgj6Cw@mail.gmail.com>
Message-ID: <33095823FD21DF429B481B5163264B7950AC4FB5F3@VMBX102.ihostexchange.net>

Here's the log with siptrace.
http://pastebin.freeswitch.org/16424

Here's the content of transfer.js:
   session.answer();
   if (session.ready())
   {
       session.execute("bridge", "{ignore_early_media=true}sofia/gateway/broadvoice/2223334444");
   }

________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins [msc at freeswitch.org]
Sent: Friday, May 27, 2011 6:21 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Transferring to a cell phone

Something is not right with the outbound leg of the call. It looks like there is an immediate hangup after the b leg answers. Get a siptrace of that traffic and look to see what is causing the hangup.

-MC

On Fri, May 27, 2011 at 2:27 PM, Yungwei Chen <yungwei at resolvity.com<mailto:yungwei at resolvity.com>> wrote:
The whole picture looks like the following:
A land-line<-->SIP provider A<-->FS<-->SIP provider A<-->a cell phone

Is there any configuration setting that needs to be enabled or set differently so that those 2 endpoints can talk to each other?
Btw, the same scenario works with Asterisk so my SIP provider shouldn't be the problem.

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Yungwei Chen
Sent: Friday, May 27, 2011 11:34 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Transferring to a cell phone

Here's the debug log, http://pastebin.freeswitch.org/16398

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
Sent: Friday, May 27, 2011 10:08 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Transferring to a cell phone

You need to put a debug log on pastebin.freeswitch.org<http://pastebin.freeswitch.org>. Be sure to use "FreeSWITCH Log" for the syntax highlighting.
-MC
On Fri, May 27, 2011 at 7:41 AM, Yungwei Chen <yungwei at resolvity.com<mailto:yungwei at resolvity.com>> wrote:
No.

I just verified that making an outbound call to my cell phone still works.
I even recorded the session just to make sure, and in the recording I hear things both ways.
I first dial 9911 from my SIP client (behind freeswitch), and this leads to the javascript program below.
  session.answer();
  var new_session = new Session("{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");
  session.hangup(16); // disconnects the session between the SIP client and freeswitch
  new_session.execute("record_session", "/tmp/call-from-js-to-cell.wav");
  while (new_session.ready())
  {
      new_session.streamFile("/path/to/local/wav/file");
  }

Now back to the case I'm having problem with.
In this case, I first make a call from a landline to freeswitch through my sip provider, and then a javascript program takes over.
I want to transfer the call to a cell phone so that the landline and the cell phone can communicate with each other.
Here's the javascript program:
  session.answer();

  if (session.ready())
  {
      session.execute("bridge", "{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");
  }

So what am I missing here?

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of mazilo
Sent: Friday, May 27, 2011 7:28 AM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Transferring to a cell phone


Yungwei Chen wrote:
>
> Thanks for your reply. Using bridge fixed the problem. But I cannot hear
> anything both ways. Any idea?
After switching to using bridge function, does this also happen when you
make an outbound call to your cell phone using your javascript?

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Transferring-to-a-cell-phone-tp6408246p6410887.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

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

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


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Wed Jun  1 22:44:45 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 1 Jun 2011 13:44:45 -0500
Subject: [Freeswitch-users] FS vs provider's Sonus DTMF (double digit)
	problem
In-Reply-To: <000001cc2062$3ae1c460$b0a54d20$@com>
References: <000001cc2062$3ae1c460$b0a54d20$@com>
Message-ID: <BANLkTinjOkqu16EjOC18+xwTm1BTyyG0Rw@mail.gmail.com>

update to latest git, I added suport to ignore INFO unless you
explicitly configure it.


On Wed, Jun 1, 2011 at 8:45 AM, Dmitry Saratsky <freeswitch at simpot.com> wrote:
> Hi All,
>
>
>
> I'm suffering from dtmf "double digit" problem for incoming calls that
> destined?to IVR on my FS.
>
> I tryed to identify and workaround the provlem with no success...
>
> I have already tryed to change dtmf-mode, use sonus-fixup etc with no
> luck...
>
> I have: FreeSWITCH Version 1.0.head (git-1086cba 2011-05-23 22-51-43 -0500)
>
> I would like to ask you some questions and I'm attaching some relevant
> output (see below)
>
>
>
> 1. According to output (see below), is it right to say, that my provider
> sends me DTMF events both in "DTMF INFO" and "RTP DTMF - rfc2833" way?
>
> 2. If so, may be this is the reason for "double digit" I suffer?
>
> 3. If so, can I filter incoming "DTMF INFO" events in FS?
>
> 4. If so, do FS have some workaround for this?
>
>
>
> Any ideas?
>
>
>
> Thanks,
>
> Dmitry.
>
>
>
> -------------------------------------------------
>
>
>
> FreeSWITCH Version 1.0.head (git-1086cba 2011-05-23 22-51-43 -0500)
>
> -------
>
>
>
> <profile name="providers">
> ? <gateways>
> ??? <X-PRE-PROCESS cmd="include" data="providers/*.xml"/>
> ? </gateways>
>
> ? <aliases>
> ? </aliases>
>
> ? <domains>
> ??? <domain name="all" alias="false" parse="true"/>
> ? </domains>
>
> ? <settings>
> ??? <param name="debug" value="0"/>
> ??? <param name="shutdown-on-fail" value="true"/>
> ??? <param name="auth-calls" value="false"/>
> ??? <param name="sip-trace" value="no"/>
> ??? <param name="disable-register" value="true"/>
> ??? <param name="rfc2833-pt" value="101"/>
> ??? <param name="dialplan" value="XML"/>
> ??? <param name="context" value="providers"/>
> ??? <param name="apply-inbound-acl" value="013SONUS-IPS"/>
> ??? <param name="hold-music" value="$${hold_music}"/>
> ??? <param name="stun-enabled" value="false"/>
> ??? <param name="stun-auto-disable" value="true"/>
> ??? <param name="manage-presence" value="false"/>
> ??? <param name="nonce-ttl" value="60"/>
> ??? <param name="dtmf-type" value="rfc2833"/>
> ??? <param name="dtmf-duration" value="2000"/>
>
> ??? <param name="rtp-timer-name" value="soft"/>
> ??? <param name="rtp-timeout-sec" value="50"/>
> ??? <param name="rtp-hold-timeout-sec" value="300"/>
>
> ??? <param name="sip-port" value="$${sip_port}"/>
>
> ??? <param name="rtp-ip" value="$${rtp_ip}"/>
> ??? <param name="sip-ip" value="$${ext_ip}"/>
>
> ??? <param name="inbound-codec-prefs" value="PCMU,PCMA"/>
> ??? <param name="outbound-codec-prefs" value="PCMU,PCMA"/>
> ??? <param name="codec-prefs" value="PCMU,PCMA"/>
> ??? <param name="inbound-late-negotiation" value="false"/>
> ??? <param name="inbound-codec-negotiation" value="scrooge"/>
> ??? <param name="disable-transcoding" value="false"/>
>
> ? </settings>
> </profile>
>
> -------
>
>
>
> 2011-06-01 10:21:30.585146 [DEBUG] sofia.c:6236 INFO DTMF(0)
> 2011-06-01 10:21:30.585146 [DEBUG] sofia.c:6310 dispatched freeswitch event
> for INFO
> 2011-06-01 10:21:30.604196 [DEBUG] switch_ivr_play_say.c:1649 done playing
> file
> 2011-06-01 10:21:30.604196 [DEBUG] switch_ivr_menu.c:343 waiting for 2/3
> digits t/o 2000
> 2011-06-01 10:21:30.845091 [DEBUG] sofia.c:6207 Bad signal
> 2011-06-01 10:21:30.845091 [DEBUG] sofia.c:6310 dispatched freeswitch event
> for INFO
> 2011-06-01 10:21:30.966061 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF 0:4360
> 2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:390 digits '00'
> 2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:484 action regex [00]
> [/^(20[1-2])$/] [0]
> 2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:484 action regex [00]
> [/^[3-9*#]$/] [0]
> 2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:574 IVR menu
> 'main_ivr_heb' caught invalid input '00'
>
>
>
> -------
>
> 2011-06-01 10:23:23.902331 [DEBUG] sofia.c:6236 INFO DTMF(7)
> 2011-06-01 10:23:23.902331 [DEBUG] sofia.c:6310 dispatched freeswitch event
> for INFO
> 2011-06-01 10:23:23.922329 [DEBUG] switch_ivr_play_say.c:1649 done playing
> file
> 2011-06-01 10:23:23.922329 [DEBUG] switch_ivr_menu.c:343 waiting for 2/3
> digits t/o 2000
> 2011-06-01 10:23:23.942304 [DEBUG] sofia.c:6207 Bad signal
> 2011-06-01 10:23:23.942304 [DEBUG] sofia.c:6310 dispatched freeswitch event
> for INFO
> 2011-06-01 10:23:24.083260 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF 7:2760
> 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:390 digits '77'
> 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:484 action regex [77]
> [/^(20[1-2])$/] [0]
> 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:484 action regex [77]
> [/^[3-9*#]$/] [0]
> 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:574 IVR menu
> 'main_ivr_heb' caught invalid input '77'
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From freeswitch at aastral.net  Thu Jun  2 00:10:18 2011
From: freeswitch at aastral.net (Bill W.)
Date: Wed, 01 Jun 2011 16:10:18 -0400
Subject: [Freeswitch-users] intermittent EXCHANGE_ROUTING_ERROR
Message-ID: <1QRrkF-0007cj-IK@mail.aastral.net>

Hey All,

About once or twice a day, I'm getting an EXCHANGE_ROUTING_ERROR failure.

I've enabled debugging, including logging SIP messages, and can't seem
to find the cause of the error.


Freeswitch goes from setting variables to exchange routing error:
switch_channel.c:935 EXPORTING[export_vars]
to
switch_ivr_originate.c:2447 Cannot create outgoing channel of type
[sofia] cause: [EXCHANGE_ROUTING_ERROR]


Whenever that happens, "switch_channel.c:816 New Channel" is never
called and I don't see any outbound SIP messages to the IP in question.

So if no new channel is created, how can there be a routing error?

Any insight or advice on how to debug this would be greatly appreciated.

Thanks!




From anthony.minessale at gmail.com  Thu Jun  2 00:36:22 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 1 Jun 2011 15:36:22 -0500
Subject: [Freeswitch-users] intermittent EXCHANGE_ROUTING_ERROR
In-Reply-To: <1QRrkF-0007cj-IK@mail.aastral.net>
References: <1QRrkF-0007cj-IK@mail.aastral.net>
Message-ID: <BANLkTikc5bMokDY6__miju=rOUmsLc7kNg@mail.gmail.com>

routing loop, its transferring more that 73 times inside the dialplan

On Wed, Jun 1, 2011 at 3:10 PM, Bill W. <freeswitch at aastral.net> wrote:
> Hey All,
>
> About once or twice a day, I'm getting an EXCHANGE_ROUTING_ERROR failure.
>
> I've enabled debugging, including logging SIP messages, and can't seem
> to find the cause of the error.
>
>
> Freeswitch goes from setting variables to exchange routing error:
> switch_channel.c:935 EXPORTING[export_vars]
> to
> switch_ivr_originate.c:2447 Cannot create outgoing channel of type
> [sofia] cause: [EXCHANGE_ROUTING_ERROR]
>
>
> Whenever that happens, "switch_channel.c:816 New Channel" is never
> called and I don't see any outbound SIP messages to the IP in question.
>
> So if no new channel is created, how can there be a routing error?
>
> Any insight or advice on how to debug this would be greatly appreciated.
>
> Thanks!
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From msc at freeswitch.org  Thu Jun  2 00:42:51 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 1 Jun 2011 13:42:51 -0700
Subject: [Freeswitch-users] ClueCon 2011 News and Notes
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59DE5494FE@cooper>
References: <00000130481a3868-69cf136f-c857-407d-90c0-e75120e27c52-000000@email.amazonses.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DE5494FE@cooper>
Message-ID: <BANLkTikEUdt69NTcP+gpyJL+x4LJfwBTeA@mail.gmail.com>

Peter,

Can you give me the names and contact information of everyone in your party?

Thanks!
-MC

On Wed, Jun 1, 2011 at 2:06 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> I booked everything today, for me and a collegue ? I?m really looking
> forward to it!
>
>
>
> /Peter
>
>
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *marketing at cluecon.com
> *Skickat:* den 1 juni 2011 00:09
> *Till:* freeswitch-users at lists.freeswitch.org
> *?mne:* [Freeswitch-users] ClueCon 2011 News and Notes
>
>
> ClueCon 2011 Reminder: Extra Chances to Win
>
> We wanted to remind everyone that you still have one more day to book
> ClueCon 2011 and receive FOUR entries into big giveaways for the coolest
> laptop and iPad <http://www.cluecon.com/coolest>. (Promotion ends on June
> 1.) Will you be the big winner? Book now and increase your chances!
>
> For those wanting to know more about what's happening at ClueCon this year
> we've posted our preliminary schedule <http://www.cluecon.com/schedule>.
> There are some great talks planned and, of course, a few surprises. Be sure
> to check back often to see the latest speakers and sponsors.
>
> Visit us at http://www.cluecon.com for more information, or call us at
> 877.742.CLUE (2583) to inquire about registration, sponsorships, or any
> other questions you may have. See you in August!
>
> The ClueCon Team
> http://www.cluecon.com
> 877.742.CLUE
>
> [image: Bild som tagits bort av avs?ndaren.] <http://www.cluecon.com>
>
> !DSPAM:4de567b932761932211873!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Jun  2 01:32:32 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 1 Jun 2011 14:32:32 -0700
Subject: [Freeswitch-users] Transferring to a cell phone
In-Reply-To: <33095823FD21DF429B481B5163264B7950AC4FB5F3@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950AC3AC74C@VMBX102.ihostexchange.net>
	<0454EB31-E9EA-4594-9FDD-29D69F5B6126@ipeva.fr>
	<33095823FD21DF429B481B5163264B7950AC3AC7E2@VMBX102.ihostexchange.net>
	<1306499299750-6410887.post@n2.nabble.com>
	<33095823FD21DF429B481B5163264B7950AC3AC8B0@VMBX102.ihostexchange.net>
	<BANLkTikrgnaTDL9+WtXh_4RAWTFnmja71Q@mail.gmail.com>
	<33095823FD21DF429B481B5163264B7950AC3AC938@VMBX102.ihostexchange.net>
	<33095823FD21DF429B481B5163264B7950AC3ACA1B@VMBX102.ihostexchange.net>
	<BANLkTi=4fMHUkHjoCZ8exuw6Qgg5Pgj6Cw@mail.gmail.com>
	<33095823FD21DF429B481B5163264B7950AC4FB5F3@VMBX102.ihostexchange.net>
Message-ID: <BANLkTimesjA5yzOFT+oVWSZzLPDVXZofrw@mail.gmail.com>

I'm not really sure what's happening here. The first BYE message that I see
comes in at line #548 from the provider side of what I assume is the A leg.

For now I would try using a very simple bridge app right in the public
context and avoid all the javascript so that you can get the simplest
scenario. SImply route the A leg to the bridge app and bridge it to the
broadvoice gateway:
<extension name="route DID">
  <condition field="destination_number"
expression="^(INCOMING_DID_NUMBER)$">
    <action application="bridge"
data="sofia/gateway/broadvoice/2223334444"/>
  </condition>
</extension>

A call coming in from one SIP endpoint and being bridged to another SIP
endpoint is pretty much the core and fundamental functionality of
FreeSWITCH, so I'm thinking that the javascript is doing something
unintended.

-MC

On Wed, Jun 1, 2011 at 9:52 AM, Yungwei Chen <yungwei at resolvity.com> wrote:

>  Here's the log with siptrace.
> http://pastebin.freeswitch.org/16424
>
> Here's the content of transfer.js:
>    session.answer();
>    if (session.ready())
>    {
>        session.execute("bridge",
> "{ignore_early_media=true}sofia/gateway/broadvoice/2223334444");
>    }
>
>  ------------------------------
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
> Collins [msc at freeswitch.org]
> *Sent:* Friday, May 27, 2011 6:21 PM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Transferring to a cell phone
>
>   Something is not right with the outbound leg of the call. It looks like
> there is an immediate hangup after the b leg answers. Get a siptrace of that
> traffic and look to see what is causing the hangup.
>
>  -MC
>
> On Fri, May 27, 2011 at 2:27 PM, Yungwei Chen <yungwei at resolvity.com>wrote:
>
>>  The whole picture looks like the following:
>>
>> A land-line<-->SIP provider A<-->FS<-->SIP provider A<-->a cell phone
>>
>>
>>
>> Is there any configuration setting that needs to be enabled or set
>> differently so that those 2 endpoints can talk to each other?
>>
>> Btw, the same scenario works with Asterisk so my SIP provider shouldn't be
>> the problem.
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Yungwei
>> Chen
>> *Sent:* Friday, May 27, 2011 11:34 AM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] Transferring to a cell phone
>>
>>
>>
>> Here's the debug log, http://pastebin.freeswitch.org/16398
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
>> Collins
>> *Sent:* Friday, May 27, 2011 10:08 AM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] Transferring to a cell phone
>>
>>
>>
>> You need to put a debug log on pastebin.freeswitch.org. Be sure to use
>> "FreeSWITCH Log" for the syntax highlighting.
>>
>> -MC
>>
>> On Fri, May 27, 2011 at 7:41 AM, Yungwei Chen <yungwei at resolvity.com>
>> wrote:
>>
>> No.
>>
>> I just verified that making an outbound call to my cell phone still works.
>> I even recorded the session just to make sure, and in the recording I hear
>> things both ways.
>> I first dial 9911 from my SIP client (behind freeswitch), and this leads
>> to the javascript program below.
>>   session.answer();
>>
>>   var new_session = new
>> Session("{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");
>>
>>   session.hangup(16); // disconnects the session between the SIP client
>> and freeswitch
>>   new_session.execute("record_session", "/tmp/call-from-js-to-cell.wav");
>>   while (new_session.ready())
>>   {
>>       new_session.streamFile("/path/to/local/wav/file");
>>   }
>>
>> Now back to the case I'm having problem with.
>> In this case, I first make a call from a landline to freeswitch through my
>> sip provider, and then a javascript program takes over.
>> I want to transfer the call to a cell phone so that the landline and the
>> cell phone can communicate with each other.
>> Here's the javascript program:
>>   session.answer();
>>
>>   if (session.ready())
>>   {
>>       session.execute("bridge",
>> "{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");
>>   }
>>
>> So what am I missing here?
>>
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of mazilo
>> Sent: Friday, May 27, 2011 7:28 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Transferring to a cell phone
>>
>>
>> Yungwei Chen wrote:
>> >
>> > Thanks for your reply. Using bridge fixed the problem. But I cannot hear
>> > anything both ways. Any idea?
>> After switching to using bridge function, does this also happen when you
>> make an outbound call to your cell phone using your javascript?
>>
>> -----
>> FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3
>> Watts of electricity.
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/Transferring-to-a-cell-phone-tp6408246p6410887.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From freeswitch at aastral.net  Thu Jun  2 01:38:14 2011
From: freeswitch at aastral.net (Bill W.)
Date: Wed, 01 Jun 2011 17:38:14 -0400
Subject: [Freeswitch-users] intermittent EXCHANGE_ROUTING_ERROR
In-Reply-To: <BANLkTikc5bMokDY6__miju=rOUmsLc7kNg@mail.gmail.com>
References: <1QRrkF-0007cj-IK@mail.aastral.net>
	<BANLkTikc5bMokDY6__miju=rOUmsLc7kNg@mail.gmail.com>
Message-ID: <1QRt7M-0002Zm-GE@mail.aastral.net>

Hey Anthony,

Thanks very much for your input.  Can you recommend a way I can prove
that to myself?  Is there some sort of logging to turn on to see that?

The reason I ask is that for this instance I receive a call in on one
sofia profile and then use the bridge command to send it out another
sofia profile. I don't use the transfer application to complete the call.



On 6/1/11 4:36 PM, Anthony Minessale wrote:
> routing loop, its transferring more that 73 times inside the dialplan
> 
> On Wed, Jun 1, 2011 at 3:10 PM, Bill W. <freeswitch at aastral.net> wrote:
>> Hey All,
>>
>> About once or twice a day, I'm getting an EXCHANGE_ROUTING_ERROR failure.
>>
>> I've enabled debugging, including logging SIP messages, and can't seem
>> to find the cause of the error.
>>
>>
>> Freeswitch goes from setting variables to exchange routing error:
>> switch_channel.c:935 EXPORTING[export_vars]
>> to
>> switch_ivr_originate.c:2447 Cannot create outgoing channel of type
>> [sofia] cause: [EXCHANGE_ROUTING_ERROR]
>>
>>
>> Whenever that happens, "switch_channel.c:816 New Channel" is never
>> called and I don't see any outbound SIP messages to the IP in question.
>>
>> So if no new channel is created, how can there be a routing error?
>>
>> Any insight or advice on how to debug this would be greatly appreciated.
>>
>> Thanks!
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> 
> 


From msc at freeswitch.org  Thu Jun  2 01:46:31 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 1 Jun 2011 14:46:31 -0700
Subject: [Freeswitch-users] intermittent EXCHANGE_ROUTING_ERROR
In-Reply-To: <1QRt7M-0002Zm-GE@mail.aastral.net>
References: <1QRrkF-0007cj-IK@mail.aastral.net>
	<BANLkTikc5bMokDY6__miju=rOUmsLc7kNg@mail.gmail.com>
	<1QRt7M-0002Zm-GE@mail.aastral.net>
Message-ID: <BANLkTikXJrxuwUXAd2kYFCiDQOLGQgw9dg@mail.gmail.com>

Well, if it's inside the dialplan you'll see it in the log. You may want to
rotate logs frequently while tracking this down. It might also help if you
enable uuid logging so that you can filter down to just a single call's log
lines. You may also wish to examine the XML CDRs for these calls. I'd be
curious to see what's in the call flows.

-MC

On Wed, Jun 1, 2011 at 2:38 PM, Bill W. <freeswitch at aastral.net> wrote:

> Hey Anthony,
>
> Thanks very much for your input.  Can you recommend a way I can prove
> that to myself?  Is there some sort of logging to turn on to see that?
>
> The reason I ask is that for this instance I receive a call in on one
> sofia profile and then use the bridge command to send it out another
> sofia profile. I don't use the transfer application to complete the call.
>
>
>
> On 6/1/11 4:36 PM, Anthony Minessale wrote:
> > routing loop, its transferring more that 73 times inside the dialplan
> >
> > On Wed, Jun 1, 2011 at 3:10 PM, Bill W. <freeswitch at aastral.net> wrote:
> >> Hey All,
> >>
> >> About once or twice a day, I'm getting an EXCHANGE_ROUTING_ERROR
> failure.
> >>
> >> I've enabled debugging, including logging SIP messages, and can't seem
> >> to find the cause of the error.
> >>
> >>
> >> Freeswitch goes from setting variables to exchange routing error:
> >> switch_channel.c:935 EXPORTING[export_vars]
> >> to
> >> switch_ivr_originate.c:2447 Cannot create outgoing channel of type
> >> [sofia] cause: [EXCHANGE_ROUTING_ERROR]
> >>
> >>
> >> Whenever that happens, "switch_channel.c:816 New Channel" is never
> >> called and I don't see any outbound SIP messages to the IP in question.
> >>
> >> So if no new channel is created, how can there be a routing error?
> >>
> >> Any insight or advice on how to debug this would be greatly appreciated.
> >>
> >> Thanks!
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From freeswitch at simpot.com  Thu Jun  2 01:47:29 2011
From: freeswitch at simpot.com (Dmitry Saratsky)
Date: Thu, 2 Jun 2011 00:47:29 +0300
Subject: [Freeswitch-users] FS vs provider's Sonus DTMF (double
	digit)	problem
In-Reply-To: <BANLkTinjOkqu16EjOC18+xwTm1BTyyG0Rw@mail.gmail.com>
References: <000001cc2062$3ae1c460$b0a54d20$@com>
	<BANLkTinjOkqu16EjOC18+xwTm1BTyyG0Rw@mail.gmail.com>
Message-ID: <000001cc20a5$817a7280$846f5780$@com>

Thanks man,

It is working great now!!!

Thanks again.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: 01 Jun 2011 21:45
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FS vs provider's Sonus DTMF (double digit)
problem

update to latest git, I added suport to ignore INFO unless you
explicitly configure it.


On Wed, Jun 1, 2011 at 8:45 AM, Dmitry Saratsky <freeswitch at simpot.com>
wrote:
> Hi All,
>
>
>
> I'm suffering from dtmf "double digit" problem for incoming calls that
> destined?to IVR on my FS.
>
> I tryed to identify and workaround the provlem with no success...
>
> I have already tryed to change dtmf-mode, use sonus-fixup etc with no
> luck...
>
> I have: FreeSWITCH Version 1.0.head (git-1086cba 2011-05-23 22-51-43
-0500)
>
> I would like to ask you some questions and I'm attaching some relevant
> output (see below)
>
>
>
> 1. According to output (see below), is it right to say, that my provider
> sends me DTMF events both in "DTMF INFO" and "RTP DTMF - rfc2833" way?
>
> 2. If so, may be this is the reason for "double digit" I suffer?
>
> 3. If so, can I filter incoming "DTMF INFO" events in FS?
>
> 4. If so, do FS have some workaround for this?
>
>
>
> Any ideas?
>
>
>
> Thanks,
>
> Dmitry.
>
>
>
> -------------------------------------------------
>
>
>
> FreeSWITCH Version 1.0.head (git-1086cba 2011-05-23 22-51-43 -0500)
>
> -------
>
>
>
> <profile name="providers">
> ? <gateways>
> ??? <X-PRE-PROCESS cmd="include" data="providers/*.xml"/>
> ? </gateways>
>
> ? <aliases>
> ? </aliases>
>
> ? <domains>
> ??? <domain name="all" alias="false" parse="true"/>
> ? </domains>
>
> ? <settings>
> ??? <param name="debug" value="0"/>
> ??? <param name="shutdown-on-fail" value="true"/>
> ??? <param name="auth-calls" value="false"/>
> ??? <param name="sip-trace" value="no"/>
> ??? <param name="disable-register" value="true"/>
> ??? <param name="rfc2833-pt" value="101"/>
> ??? <param name="dialplan" value="XML"/>
> ??? <param name="context" value="providers"/>
> ??? <param name="apply-inbound-acl" value="013SONUS-IPS"/>
> ??? <param name="hold-music" value="$${hold_music}"/>
> ??? <param name="stun-enabled" value="false"/>
> ??? <param name="stun-auto-disable" value="true"/>
> ??? <param name="manage-presence" value="false"/>
> ??? <param name="nonce-ttl" value="60"/>
> ??? <param name="dtmf-type" value="rfc2833"/>
> ??? <param name="dtmf-duration" value="2000"/>
>
> ??? <param name="rtp-timer-name" value="soft"/>
> ??? <param name="rtp-timeout-sec" value="50"/>
> ??? <param name="rtp-hold-timeout-sec" value="300"/>
>
> ??? <param name="sip-port" value="$${sip_port}"/>
>
> ??? <param name="rtp-ip" value="$${rtp_ip}"/>
> ??? <param name="sip-ip" value="$${ext_ip}"/>
>
> ??? <param name="inbound-codec-prefs" value="PCMU,PCMA"/>
> ??? <param name="outbound-codec-prefs" value="PCMU,PCMA"/>
> ??? <param name="codec-prefs" value="PCMU,PCMA"/>
> ??? <param name="inbound-late-negotiation" value="false"/>
> ??? <param name="inbound-codec-negotiation" value="scrooge"/>
> ??? <param name="disable-transcoding" value="false"/>
>
> ? </settings>
> </profile>
>
> -------
>
>
>
> 2011-06-01 10:21:30.585146 [DEBUG] sofia.c:6236 INFO DTMF(0)
> 2011-06-01 10:21:30.585146 [DEBUG] sofia.c:6310 dispatched freeswitch
event
> for INFO
> 2011-06-01 10:21:30.604196 [DEBUG] switch_ivr_play_say.c:1649 done playing
> file
> 2011-06-01 10:21:30.604196 [DEBUG] switch_ivr_menu.c:343 waiting for 2/3
> digits t/o 2000
> 2011-06-01 10:21:30.845091 [DEBUG] sofia.c:6207 Bad signal
> 2011-06-01 10:21:30.845091 [DEBUG] sofia.c:6310 dispatched freeswitch
event
> for INFO
> 2011-06-01 10:21:30.966061 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF 0:4360
> 2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:390 digits '00'
> 2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:484 action regex [00]
> [/^(20[1-2])$/] [0]
> 2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:484 action regex [00]
> [/^[3-9*#]$/] [0]
> 2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:574 IVR menu
> 'main_ivr_heb' caught invalid input '00'
>
>
>
> -------
>
> 2011-06-01 10:23:23.902331 [DEBUG] sofia.c:6236 INFO DTMF(7)
> 2011-06-01 10:23:23.902331 [DEBUG] sofia.c:6310 dispatched freeswitch
event
> for INFO
> 2011-06-01 10:23:23.922329 [DEBUG] switch_ivr_play_say.c:1649 done playing
> file
> 2011-06-01 10:23:23.922329 [DEBUG] switch_ivr_menu.c:343 waiting for 2/3
> digits t/o 2000
> 2011-06-01 10:23:23.942304 [DEBUG] sofia.c:6207 Bad signal
> 2011-06-01 10:23:23.942304 [DEBUG] sofia.c:6310 dispatched freeswitch
event
> for INFO
> 2011-06-01 10:23:24.083260 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF 7:2760
> 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:390 digits '77'
> 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:484 action regex [77]
> [/^(20[1-2])$/] [0]
> 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:484 action regex [77]
> [/^[3-9*#]$/] [0]
> 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:574 IVR menu
> 'main_ivr_heb' caught invalid input '77'
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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

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



From freeswitch at simpot.com  Thu Jun  2 02:36:29 2011
From: freeswitch at simpot.com (Dmitry Saratsky)
Date: Thu, 2 Jun 2011 01:36:29 +0300
Subject: [Freeswitch-users] FS vs provider's Sonus DTMF (double
	digit)	problem
In-Reply-To: <BANLkTinjOkqu16EjOC18+xwTm1BTyyG0Rw@mail.gmail.com>
References: <000001cc2062$3ae1c460$b0a54d20$@com>
	<BANLkTinjOkqu16EjOC18+xwTm1BTyyG0Rw@mail.gmail.com>
Message-ID: <000101cc20ac$598c1c40$0ca454c0$@com>

One more question:

Is this behavior not just may be called as "unusual", but definitely
prohibited by protocol description/rfc/etc? bcz I'm thinking to make
mega-BOOM to my provider, but I have to know if it "strange" or "prohibited"
behavior...

Thanks.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: 01 Jun 2011 21:45
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FS vs provider's Sonus DTMF (double digit)
problem

update to latest git, I added suport to ignore INFO unless you
explicitly configure it.


On Wed, Jun 1, 2011 at 8:45 AM, Dmitry Saratsky <freeswitch at simpot.com>
wrote:
> Hi All,
>
>
>
> I'm suffering from dtmf "double digit" problem for incoming calls that
> destined?to IVR on my FS.
>
> I tryed to identify and workaround the provlem with no success...
>
> I have already tryed to change dtmf-mode, use sonus-fixup etc with no
> luck...
>
> I have: FreeSWITCH Version 1.0.head (git-1086cba 2011-05-23 22-51-43
-0500)
>
> I would like to ask you some questions and I'm attaching some relevant
> output (see below)
>
>
>
> 1. According to output (see below), is it right to say, that my provider
> sends me DTMF events both in "DTMF INFO" and "RTP DTMF - rfc2833" way?
>
> 2. If so, may be this is the reason for "double digit" I suffer?
>
> 3. If so, can I filter incoming "DTMF INFO" events in FS?
>
> 4. If so, do FS have some workaround for this?
>
>
>
> Any ideas?
>
>
>
> Thanks,
>
> Dmitry.
>
>
>
> -------------------------------------------------
>
>
>
> FreeSWITCH Version 1.0.head (git-1086cba 2011-05-23 22-51-43 -0500)
>
> -------
>
>
>
> <profile name="providers">
> ? <gateways>
> ??? <X-PRE-PROCESS cmd="include" data="providers/*.xml"/>
> ? </gateways>
>
> ? <aliases>
> ? </aliases>
>
> ? <domains>
> ??? <domain name="all" alias="false" parse="true"/>
> ? </domains>
>
> ? <settings>
> ??? <param name="debug" value="0"/>
> ??? <param name="shutdown-on-fail" value="true"/>
> ??? <param name="auth-calls" value="false"/>
> ??? <param name="sip-trace" value="no"/>
> ??? <param name="disable-register" value="true"/>
> ??? <param name="rfc2833-pt" value="101"/>
> ??? <param name="dialplan" value="XML"/>
> ??? <param name="context" value="providers"/>
> ??? <param name="apply-inbound-acl" value="013SONUS-IPS"/>
> ??? <param name="hold-music" value="$${hold_music}"/>
> ??? <param name="stun-enabled" value="false"/>
> ??? <param name="stun-auto-disable" value="true"/>
> ??? <param name="manage-presence" value="false"/>
> ??? <param name="nonce-ttl" value="60"/>
> ??? <param name="dtmf-type" value="rfc2833"/>
> ??? <param name="dtmf-duration" value="2000"/>
>
> ??? <param name="rtp-timer-name" value="soft"/>
> ??? <param name="rtp-timeout-sec" value="50"/>
> ??? <param name="rtp-hold-timeout-sec" value="300"/>
>
> ??? <param name="sip-port" value="$${sip_port}"/>
>
> ??? <param name="rtp-ip" value="$${rtp_ip}"/>
> ??? <param name="sip-ip" value="$${ext_ip}"/>
>
> ??? <param name="inbound-codec-prefs" value="PCMU,PCMA"/>
> ??? <param name="outbound-codec-prefs" value="PCMU,PCMA"/>
> ??? <param name="codec-prefs" value="PCMU,PCMA"/>
> ??? <param name="inbound-late-negotiation" value="false"/>
> ??? <param name="inbound-codec-negotiation" value="scrooge"/>
> ??? <param name="disable-transcoding" value="false"/>
>
> ? </settings>
> </profile>
>
> -------
>
>
>
> 2011-06-01 10:21:30.585146 [DEBUG] sofia.c:6236 INFO DTMF(0)
> 2011-06-01 10:21:30.585146 [DEBUG] sofia.c:6310 dispatched freeswitch
event
> for INFO
> 2011-06-01 10:21:30.604196 [DEBUG] switch_ivr_play_say.c:1649 done playing
> file
> 2011-06-01 10:21:30.604196 [DEBUG] switch_ivr_menu.c:343 waiting for 2/3
> digits t/o 2000
> 2011-06-01 10:21:30.845091 [DEBUG] sofia.c:6207 Bad signal
> 2011-06-01 10:21:30.845091 [DEBUG] sofia.c:6310 dispatched freeswitch
event
> for INFO
> 2011-06-01 10:21:30.966061 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF 0:4360
> 2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:390 digits '00'
> 2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:484 action regex [00]
> [/^(20[1-2])$/] [0]
> 2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:484 action regex [00]
> [/^[3-9*#]$/] [0]
> 2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:574 IVR menu
> 'main_ivr_heb' caught invalid input '00'
>
>
>
> -------
>
> 2011-06-01 10:23:23.902331 [DEBUG] sofia.c:6236 INFO DTMF(7)
> 2011-06-01 10:23:23.902331 [DEBUG] sofia.c:6310 dispatched freeswitch
event
> for INFO
> 2011-06-01 10:23:23.922329 [DEBUG] switch_ivr_play_say.c:1649 done playing
> file
> 2011-06-01 10:23:23.922329 [DEBUG] switch_ivr_menu.c:343 waiting for 2/3
> digits t/o 2000
> 2011-06-01 10:23:23.942304 [DEBUG] sofia.c:6207 Bad signal
> 2011-06-01 10:23:23.942304 [DEBUG] sofia.c:6310 dispatched freeswitch
event
> for INFO
> 2011-06-01 10:23:24.083260 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF 7:2760
> 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:390 digits '77'
> 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:484 action regex [77]
> [/^(20[1-2])$/] [0]
> 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:484 action regex [77]
> [/^[3-9*#]$/] [0]
> 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:574 IVR menu
> 'main_ivr_heb' caught invalid input '77'
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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

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



From msc at freeswitch.org  Thu Jun  2 02:45:18 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 1 Jun 2011 15:45:18 -0700
Subject: [Freeswitch-users] PLAY_AND_GET_DIGITS dialplann application:
 set <lang> and <say_type> to SAY subcommand
In-Reply-To: <DF4F093EDA9F48BC9C688D75AB774225@abisoft.biz>
References: <DF4F093EDA9F48BC9C688D75AB774225@abisoft.biz>
Message-ID: <BANLkTikiVb3x+GjdqsPy=9_gVbeGR4ECPw@mail.gmail.com>

There is a difference between "say" and "speak". The say modules require you
to download sound files. The speak application requires at least one TTS
engine installed and running. I think you might need to go back and re-read
the say and speak entries on the wiki to make sure that you know which one
you should be using.

-MC

2011/6/1 Kozak Vladimir <vkozak at abisoft.spb.ru>

>  Hello all.
> In example for "play_and_get_digits" dialplan app I see next extension.
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits
>
> <extension name="play_and_get_digits with say">
>   <condition field="destination_number" expression="^(6500)$">
>     <action application="play_and_get_digits" data="1 1 1 3000 # say:'press one for technicial support' silence_stream://250 res \d+" />
>   </condition>
> </extension>
>
> This extension say somethin and get user DTMF.
> I need say digits and numbers on various (EN and ES) languages.
>  It's possible to specify these vars for "say" dialplan application
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_say. And it works
> correctly.
>
> Could you tell me How can I set language and say_type for tts_engine in
> "play_and_get_digits" app case?
>
> I try to use next extension, but it don't work (FS take 'es NUMBER
> 1234567'  as <text_to_say> and say all this text on English):
>  <extension name="read_extension3">
>         <condition field="destination_number" expression="^7890$">
>             <action application="answer"/>
>                 <action application="set" data="tts_engine=flite"/>
>                 <action application="set" data="tts_voice=kal"/>
>  <action application="play_and_get_digits" data="1 1 1 3000 # say:'es
> NUMBER 1234567' /usr/local/freeswitch/sounds/starpound/silence.wav
> read_result \d+" />
>             <action application="event"
> data="Event-App-Type=READ-EXECUTED,Read-Result=${read_result}"/>
>             <action application="log" data="Read-Result=${read_result}"/>
>             <action application="park"/>
>         </condition>
>     </extension>
>
> Also I try to set vars:
>                 <action application="set" data="tts_engine=cepstral"/>
>                 <action application="set" data="tts_voice=maria"/>
> <!-- ES lang -->
> and get failed result:
> EXECUTE sofia/internal/1007 at 172.26.200.250:5060 play_and_get_digits(1 1 1
> 3000 # say:'1234567' /usr/local/freeswitch/sounds/starpound/silence.wav
> read_result \d+)
> 2011-06-05 22:58:33.661014 [ERR] switch_core_speech.c:61 Invalid speech
> module [cepstral]!
> 2011-06-05 22:58:33.661014 [ERR] switch_ivr_play_say.c:2361 Invalid TTS
> module!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Thu Jun  2 03:13:02 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 1 Jun 2011 18:13:02 -0500
Subject: [Freeswitch-users] FS vs provider's Sonus DTMF (double digit)
	problem
In-Reply-To: <000101cc20ac$598c1c40$0ca454c0$@com>
References: <000001cc2062$3ae1c460$b0a54d20$@com>
	<BANLkTinjOkqu16EjOC18+xwTm1BTyyG0Rw@mail.gmail.com>
	<000101cc20ac$598c1c40$0ca454c0$@com>
Message-ID: <BANLkTinuarXu-XEJQ9Hy7HcafuYzpECGfA@mail.gmail.com>

its kind of silly but I can see some kind of rationale for it if you
are trying to allow both the signalling and the media to see dtmf for
sure but you certainly need to ignore one of them.  Technically, INFO
dtmf does not exist, the draft expired and was not adopted so it's a
rogue method now.  Everyone already uses it so i'm sure it will stay
around.  I don't think there is a real way to negotiate it however.


On Wed, Jun 1, 2011 at 5:36 PM, Dmitry Saratsky <freeswitch at simpot.com> wrote:
> One more question:
>
> Is this behavior not just may be called as "unusual", but definitely
> prohibited by protocol description/rfc/etc? bcz I'm thinking to make
> mega-BOOM to my provider, but I have to know if it "strange" or "prohibited"
> behavior...
>
> Thanks.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
> Minessale
> Sent: 01 Jun 2011 21:45
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] FS vs provider's Sonus DTMF (double digit)
> problem
>
> update to latest git, I added suport to ignore INFO unless you
> explicitly configure it.
>
>
> On Wed, Jun 1, 2011 at 8:45 AM, Dmitry Saratsky <freeswitch at simpot.com>
> wrote:
>> Hi All,
>>
>>
>>
>> I'm suffering from dtmf "double digit" problem for incoming calls that
>> destined?to IVR on my FS.
>>
>> I tryed to identify and workaround the provlem with no success...
>>
>> I have already tryed to change dtmf-mode, use sonus-fixup etc with no
>> luck...
>>
>> I have: FreeSWITCH Version 1.0.head (git-1086cba 2011-05-23 22-51-43
> -0500)
>>
>> I would like to ask you some questions and I'm attaching some relevant
>> output (see below)
>>
>>
>>
>> 1. According to output (see below), is it right to say, that my provider
>> sends me DTMF events both in "DTMF INFO" and "RTP DTMF - rfc2833" way?
>>
>> 2. If so, may be this is the reason for "double digit" I suffer?
>>
>> 3. If so, can I filter incoming "DTMF INFO" events in FS?
>>
>> 4. If so, do FS have some workaround for this?
>>
>>
>>
>> Any ideas?
>>
>>
>>
>> Thanks,
>>
>> Dmitry.
>>
>>
>>
>> -------------------------------------------------
>>
>>
>>
>> FreeSWITCH Version 1.0.head (git-1086cba 2011-05-23 22-51-43 -0500)
>>
>> -------
>>
>>
>>
>> <profile name="providers">
>> ? <gateways>
>> ??? <X-PRE-PROCESS cmd="include" data="providers/*.xml"/>
>> ? </gateways>
>>
>> ? <aliases>
>> ? </aliases>
>>
>> ? <domains>
>> ??? <domain name="all" alias="false" parse="true"/>
>> ? </domains>
>>
>> ? <settings>
>> ??? <param name="debug" value="0"/>
>> ??? <param name="shutdown-on-fail" value="true"/>
>> ??? <param name="auth-calls" value="false"/>
>> ??? <param name="sip-trace" value="no"/>
>> ??? <param name="disable-register" value="true"/>
>> ??? <param name="rfc2833-pt" value="101"/>
>> ??? <param name="dialplan" value="XML"/>
>> ??? <param name="context" value="providers"/>
>> ??? <param name="apply-inbound-acl" value="013SONUS-IPS"/>
>> ??? <param name="hold-music" value="$${hold_music}"/>
>> ??? <param name="stun-enabled" value="false"/>
>> ??? <param name="stun-auto-disable" value="true"/>
>> ??? <param name="manage-presence" value="false"/>
>> ??? <param name="nonce-ttl" value="60"/>
>> ??? <param name="dtmf-type" value="rfc2833"/>
>> ??? <param name="dtmf-duration" value="2000"/>
>>
>> ??? <param name="rtp-timer-name" value="soft"/>
>> ??? <param name="rtp-timeout-sec" value="50"/>
>> ??? <param name="rtp-hold-timeout-sec" value="300"/>
>>
>> ??? <param name="sip-port" value="$${sip_port}"/>
>>
>> ??? <param name="rtp-ip" value="$${rtp_ip}"/>
>> ??? <param name="sip-ip" value="$${ext_ip}"/>
>>
>> ??? <param name="inbound-codec-prefs" value="PCMU,PCMA"/>
>> ??? <param name="outbound-codec-prefs" value="PCMU,PCMA"/>
>> ??? <param name="codec-prefs" value="PCMU,PCMA"/>
>> ??? <param name="inbound-late-negotiation" value="false"/>
>> ??? <param name="inbound-codec-negotiation" value="scrooge"/>
>> ??? <param name="disable-transcoding" value="false"/>
>>
>> ? </settings>
>> </profile>
>>
>> -------
>>
>>
>>
>> 2011-06-01 10:21:30.585146 [DEBUG] sofia.c:6236 INFO DTMF(0)
>> 2011-06-01 10:21:30.585146 [DEBUG] sofia.c:6310 dispatched freeswitch
> event
>> for INFO
>> 2011-06-01 10:21:30.604196 [DEBUG] switch_ivr_play_say.c:1649 done playing
>> file
>> 2011-06-01 10:21:30.604196 [DEBUG] switch_ivr_menu.c:343 waiting for 2/3
>> digits t/o 2000
>> 2011-06-01 10:21:30.845091 [DEBUG] sofia.c:6207 Bad signal
>> 2011-06-01 10:21:30.845091 [DEBUG] sofia.c:6310 dispatched freeswitch
> event
>> for INFO
>> 2011-06-01 10:21:30.966061 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF 0:4360
>> 2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:390 digits '00'
>> 2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:484 action regex [00]
>> [/^(20[1-2])$/] [0]
>> 2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:484 action regex [00]
>> [/^[3-9*#]$/] [0]
>> 2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:574 IVR menu
>> 'main_ivr_heb' caught invalid input '00'
>>
>>
>>
>> -------
>>
>> 2011-06-01 10:23:23.902331 [DEBUG] sofia.c:6236 INFO DTMF(7)
>> 2011-06-01 10:23:23.902331 [DEBUG] sofia.c:6310 dispatched freeswitch
> event
>> for INFO
>> 2011-06-01 10:23:23.922329 [DEBUG] switch_ivr_play_say.c:1649 done playing
>> file
>> 2011-06-01 10:23:23.922329 [DEBUG] switch_ivr_menu.c:343 waiting for 2/3
>> digits t/o 2000
>> 2011-06-01 10:23:23.942304 [DEBUG] sofia.c:6207 Bad signal
>> 2011-06-01 10:23:23.942304 [DEBUG] sofia.c:6310 dispatched freeswitch
> event
>> for INFO
>> 2011-06-01 10:23:24.083260 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF 7:2760
>> 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:390 digits '77'
>> 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:484 action regex [77]
>> [/^(20[1-2])$/] [0]
>> 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:484 action regex [77]
>> [/^[3-9*#]$/] [0]
>> 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:574 IVR menu
>> 'main_ivr_heb' caught invalid input '77'
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From yungwei at resolvity.com  Thu Jun  2 03:28:01 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Wed, 1 Jun 2011 19:28:01 -0400
Subject: [Freeswitch-users] Unable to get events from freeswitch using java
	ESL client
Message-ID: <33095823FD21DF429B481B5163264B7950AC4FB5F7@VMBX102.ihostexchange.net>

Hi,

I am using org.freeswitch.esl.client-0.9.2 to ask freeswitch to make an outbound call in the following java class.
I want to be able to get notified when a call is not answered for whatever reason.
However, I don't get any events from freeswitch. What am I missing here?
Thanks.

public class esl implements IEslEventListener {

 /**
  * @param args
  */
 public static void main(String[] args) {
  esl e = new esl();
  Client client = new Client();
  EslMessage response = null;

  try {
   client.connect("192.168.6.18", 8021, "ClueCon", 10);
   client.addEventListener(e);

   response = client.sendSyncApiCommand("originate", "{ignore_early_media=true}sofia/gateway/broadvoice/1112223333 9886");

   CommandResponse cr = client.close();
   System.out.println("Done.");

  } catch (InboundConnectionFailure ex) {
   ex.printStackTrace();
  }
 }
 @Override
 public void backgroundJobResultReceived(EslEvent arg0) {
  System.out.println("backgroundJobResultReceived");
 }
 @Override
 public void eventReceived(EslEvent arg0) {
  System.out.println(String.format("eventReceived, event name=%s", arg0.getEventName()));
 }
}
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From freeswitch at simpot.com  Thu Jun  2 03:34:34 2011
From: freeswitch at simpot.com (Dmitry Saratsky)
Date: Thu, 2 Jun 2011 02:34:34 +0300
Subject: [Freeswitch-users] FS vs provider's Sonus DTMF (double
	digit)	problem
In-Reply-To: <BANLkTinuarXu-XEJQ9Hy7HcafuYzpECGfA@mail.gmail.com>
References: <000001cc2062$3ae1c460$b0a54d20$@com>	<BANLkTinjOkqu16EjOC18+xwTm1BTyyG0Rw@mail.gmail.com>	<000101cc20ac$598c1c40$0ca454c0$@com>
	<BANLkTinuarXu-XEJQ9Hy7HcafuYzpECGfA@mail.gmail.com>
Message-ID: <000201cc20b4$77012e70$65038b50$@com>

Ok, thanks.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: 02 Jun 2011 02:13
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FS vs provider's Sonus DTMF (double digit)
problem

its kind of silly but I can see some kind of rationale for it if you
are trying to allow both the signalling and the media to see dtmf for
sure but you certainly need to ignore one of them.  Technically, INFO
dtmf does not exist, the draft expired and was not adopted so it's a
rogue method now.  Everyone already uses it so i'm sure it will stay
around.  I don't think there is a real way to negotiate it however.


On Wed, Jun 1, 2011 at 5:36 PM, Dmitry Saratsky <freeswitch at simpot.com>
wrote:
> One more question:
>
> Is this behavior not just may be called as "unusual", but definitely
> prohibited by protocol description/rfc/etc? bcz I'm thinking to make
> mega-BOOM to my provider, but I have to know if it "strange" or
"prohibited"
> behavior...
>
> Thanks.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony
> Minessale
> Sent: 01 Jun 2011 21:45
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] FS vs provider's Sonus DTMF (double digit)
> problem
>
> update to latest git, I added suport to ignore INFO unless you
> explicitly configure it.
>
>
> On Wed, Jun 1, 2011 at 8:45 AM, Dmitry Saratsky <freeswitch at simpot.com>
> wrote:
>> Hi All,
>>
>>
>>
>> I'm suffering from dtmf "double digit" problem for incoming calls that
>> destined?to IVR on my FS.
>>
>> I tryed to identify and workaround the provlem with no success...
>>
>> I have already tryed to change dtmf-mode, use sonus-fixup etc with no
>> luck...
>>
>> I have: FreeSWITCH Version 1.0.head (git-1086cba 2011-05-23 22-51-43
> -0500)
>>
>> I would like to ask you some questions and I'm attaching some relevant
>> output (see below)
>>
>>
>>
>> 1. According to output (see below), is it right to say, that my provider
>> sends me DTMF events both in "DTMF INFO" and "RTP DTMF - rfc2833" way?
>>
>> 2. If so, may be this is the reason for "double digit" I suffer?
>>
>> 3. If so, can I filter incoming "DTMF INFO" events in FS?
>>
>> 4. If so, do FS have some workaround for this?
>>
>>
>>
>> Any ideas?
>>
>>
>>
>> Thanks,
>>
>> Dmitry.
>>
>>
>>
>> -------------------------------------------------
>>
>>
>>
>> FreeSWITCH Version 1.0.head (git-1086cba 2011-05-23 22-51-43 -0500)
>>
>> -------
>>
>>
>>
>> <profile name="providers">
>> ? <gateways>
>> ??? <X-PRE-PROCESS cmd="include" data="providers/*.xml"/>
>> ? </gateways>
>>
>> ? <aliases>
>> ? </aliases>
>>
>> ? <domains>
>> ??? <domain name="all" alias="false" parse="true"/>
>> ? </domains>
>>
>> ? <settings>
>> ??? <param name="debug" value="0"/>
>> ??? <param name="shutdown-on-fail" value="true"/>
>> ??? <param name="auth-calls" value="false"/>
>> ??? <param name="sip-trace" value="no"/>
>> ??? <param name="disable-register" value="true"/>
>> ??? <param name="rfc2833-pt" value="101"/>
>> ??? <param name="dialplan" value="XML"/>
>> ??? <param name="context" value="providers"/>
>> ??? <param name="apply-inbound-acl" value="013SONUS-IPS"/>
>> ??? <param name="hold-music" value="$${hold_music}"/>
>> ??? <param name="stun-enabled" value="false"/>
>> ??? <param name="stun-auto-disable" value="true"/>
>> ??? <param name="manage-presence" value="false"/>
>> ??? <param name="nonce-ttl" value="60"/>
>> ??? <param name="dtmf-type" value="rfc2833"/>
>> ??? <param name="dtmf-duration" value="2000"/>
>>
>> ??? <param name="rtp-timer-name" value="soft"/>
>> ??? <param name="rtp-timeout-sec" value="50"/>
>> ??? <param name="rtp-hold-timeout-sec" value="300"/>
>>
>> ??? <param name="sip-port" value="$${sip_port}"/>
>>
>> ??? <param name="rtp-ip" value="$${rtp_ip}"/>
>> ??? <param name="sip-ip" value="$${ext_ip}"/>
>>
>> ??? <param name="inbound-codec-prefs" value="PCMU,PCMA"/>
>> ??? <param name="outbound-codec-prefs" value="PCMU,PCMA"/>
>> ??? <param name="codec-prefs" value="PCMU,PCMA"/>
>> ??? <param name="inbound-late-negotiation" value="false"/>
>> ??? <param name="inbound-codec-negotiation" value="scrooge"/>
>> ??? <param name="disable-transcoding" value="false"/>
>>
>> ? </settings>
>> </profile>
>>
>> -------
>>
>>
>>
>> 2011-06-01 10:21:30.585146 [DEBUG] sofia.c:6236 INFO DTMF(0)
>> 2011-06-01 10:21:30.585146 [DEBUG] sofia.c:6310 dispatched freeswitch
> event
>> for INFO
>> 2011-06-01 10:21:30.604196 [DEBUG] switch_ivr_play_say.c:1649 done
playing
>> file
>> 2011-06-01 10:21:30.604196 [DEBUG] switch_ivr_menu.c:343 waiting for 2/3
>> digits t/o 2000
>> 2011-06-01 10:21:30.845091 [DEBUG] sofia.c:6207 Bad signal
>> 2011-06-01 10:21:30.845091 [DEBUG] sofia.c:6310 dispatched freeswitch
> event
>> for INFO
>> 2011-06-01 10:21:30.966061 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF 0:4360
>> 2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:390 digits '00'
>> 2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:484 action regex
[00]
>> [/^(20[1-2])$/] [0]
>> 2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:484 action regex
[00]
>> [/^[3-9*#]$/] [0]
>> 2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:574 IVR menu
>> 'main_ivr_heb' caught invalid input '00'
>>
>>
>>
>> -------
>>
>> 2011-06-01 10:23:23.902331 [DEBUG] sofia.c:6236 INFO DTMF(7)
>> 2011-06-01 10:23:23.902331 [DEBUG] sofia.c:6310 dispatched freeswitch
> event
>> for INFO
>> 2011-06-01 10:23:23.922329 [DEBUG] switch_ivr_play_say.c:1649 done
playing
>> file
>> 2011-06-01 10:23:23.922329 [DEBUG] switch_ivr_menu.c:343 waiting for 2/3
>> digits t/o 2000
>> 2011-06-01 10:23:23.942304 [DEBUG] sofia.c:6207 Bad signal
>> 2011-06-01 10:23:23.942304 [DEBUG] sofia.c:6310 dispatched freeswitch
> event
>> for INFO
>> 2011-06-01 10:23:24.083260 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF 7:2760
>> 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:390 digits '77'
>> 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:484 action regex
[77]
>> [/^(20[1-2])$/] [0]
>> 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:484 action regex
[77]
>> [/^[3-9*#]$/] [0]
>> 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:574 IVR menu
>> 'main_ivr_heb' caught invalid input '77'
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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

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



From bobc at devassert.com  Thu Jun  2 03:38:29 2011
From: bobc at devassert.com (Bob Coleman)
Date: Thu, 2 Jun 2011 11:38:29 +1200
Subject: [Freeswitch-users] FS vs provider's Sonus DTMF (double digit)
	problem
In-Reply-To: <BANLkTi=bwO5AUw4Py3W+4761_1w7TwFUGQ@mail.gmail.com>
References: <BANLkTi=bwO5AUw4Py3W+4761_1w7TwFUGQ@mail.gmail.com>
Message-ID: <BANLkTinT=xm8R6XSg-r95ccR=Ubrt94oVA@mail.gmail.com>

Are you using start_dtmf in the ivr?  If so try it without.

You could also go the other way and change the dtmf_type to "info" and
use start_dtmf in the ivr

Bob

On Wed, Jun 1, 2011 at 7:40 PM, Dmitry Saratsky <simpot at gmail.com> wrote:
> Hi All,
>
> I'm suffering from dtmf "double digit" problem for incoming calls that
> destined?to IVR on my FS.
> I tryed to identify and workaround the provlem with no success...
> I have already tryed to change dtmf-mode, use sonus-fixup etc with no
> luck...
> I have: FreeSWITCH Version 1.0.head (git-1086cba 2011-05-23 22-51-43 -0500)
> I would like to ask you some questions and I'm attaching some relevant
> output (see below)
>


From freeswitch at simpot.com  Thu Jun  2 03:48:39 2011
From: freeswitch at simpot.com (Dmitry Saratsky)
Date: Thu, 2 Jun 2011 02:48:39 +0300
Subject: [Freeswitch-users] FS vs provider's Sonus DTMF (double
	digit)	problem
In-Reply-To: <BANLkTinT=xm8R6XSg-r95ccR=Ubrt94oVA@mail.gmail.com>
References: <BANLkTi=bwO5AUw4Py3W+4761_1w7TwFUGQ@mail.gmail.com>
	<BANLkTinT=xm8R6XSg-r95ccR=Ubrt94oVA@mail.gmail.com>
Message-ID: <000301cc20b6$6e5f0740$4b1d15c0$@com>

Hi Bob,

Thank you for your input.
Problem already fixed, see
http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073155.html

For some reason my first post was not posted for a long time (~6 hours)...
so I posted it through my second account immediately...
Sorry for double post... I have no idea how (if it possible at all) to
remove my first cross-post now...

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bob
Coleman
Sent: 02 Jun 2011 02:38
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FS vs provider's Sonus DTMF (double digit)
problem

Are you using start_dtmf in the ivr?  If so try it without.

You could also go the other way and change the dtmf_type to "info" and
use start_dtmf in the ivr

Bob

On Wed, Jun 1, 2011 at 7:40 PM, Dmitry Saratsky <simpot at gmail.com> wrote:
> Hi All,
>
> I'm suffering from dtmf "double digit" problem for incoming calls that
> destined?to IVR on my FS.
> I tryed to identify and workaround the provlem with no success...
> I have already tryed to change dtmf-mode, use sonus-fixup etc with no
> luck...
> I have: FreeSWITCH Version 1.0.head (git-1086cba 2011-05-23 22-51-43
-0500)
> I would like to ask you some questions and I'm attaching some relevant
> output (see below)
>

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



From yungwei at resolvity.com  Thu Jun  2 04:17:53 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Wed, 1 Jun 2011 20:17:53 -0400
Subject: [Freeswitch-users] Transferring to a cell phone
In-Reply-To: <BANLkTimesjA5yzOFT+oVWSZzLPDVXZofrw@mail.gmail.com>
References: <33095823FD21DF429B481B5163264B7950AC3AC74C@VMBX102.ihostexchange.net>
	<0454EB31-E9EA-4594-9FDD-29D69F5B6126@ipeva.fr>
	<33095823FD21DF429B481B5163264B7950AC3AC7E2@VMBX102.ihostexchange.net>
	<1306499299750-6410887.post@n2.nabble.com>
	<33095823FD21DF429B481B5163264B7950AC3AC8B0@VMBX102.ihostexchange.net>
	<BANLkTikrgnaTDL9+WtXh_4RAWTFnmja71Q@mail.gmail.com>
	<33095823FD21DF429B481B5163264B7950AC3AC938@VMBX102.ihostexchange.net>
	<33095823FD21DF429B481B5163264B7950AC3ACA1B@VMBX102.ihostexchange.net>
	<BANLkTi=4fMHUkHjoCZ8exuw6Qgg5Pgj6Cw@mail.gmail.com>
	<33095823FD21DF429B481B5163264B7950AC4FB5F3@VMBX102.ihostexchange.net>,
	<BANLkTimesjA5yzOFT+oVWSZzLPDVXZofrw@mail.gmail.com>
Message-ID: <33095823FD21DF429B481B5163264B7950AC4FB5F8@VMBX102.ihostexchange.net>

Using bridge in the dialplan directly works.

I tried putting only the following line in transfer.js, but it doesn't work.
       session.execute("bridge", "sofia/gateway/broadvoice/2223334444");

Putting the following instead doesn't work either.
       session.execute("bridge", "{ignore_early_media=true}sofia/gateway/broadvoice/2223334444");

Using the following instead doesn't work either.
       session.answer();
       session.execute("bridge", "{ignore_early_media=true}sofia/gateway/broadvoice/2223334444");

Still don't understand why.

From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins [msc at freeswitch.org]
Sent: Wednesday, June 01, 2011 4:32 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Transferring to a cell phone


I'm not really sure what's happening here. The first BYE message that I see comes in at line #548 from the provider side of what I assume is the A leg.  


For now I would try using a very simple bridge app right in the public context and avoid all the javascript so that you can get the simplest scenario. SImply route the A leg to the bridge app and bridge it to the broadvoice gateway:
<extension name="route DID">
  <condition field="destination_number" expression="^(INCOMING_DID_NUMBER)$">
    <action application="bridge" data="sofia/gateway/broadvoice/2223334444"/>
  </condition>
</extension>



A call coming in from one SIP endpoint and being bridged to another SIP endpoint is pretty much the core and fundamental functionality of FreeSWITCH, so I'm thinking that the javascript is doing something unintended. 


-MC


On Wed, Jun 1, 2011 at 9:52 AM, Yungwei Chen <yungwei at resolvity.com> wrote:

Here's the log with siptrace.
http://pastebin.freeswitch.org/16424

Here's the content of transfer.js:
   session.answer();
   if (session.ready())
   {
       session.execute("bridge", "{ignore_early_media=true}sofia/gateway/broadvoice/2223334444");
   }




From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins [msc at freeswitch.org]
Sent: Friday, May 27, 2011 6:21 PM 

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Transferring to a cell phone



Something is not right with the outbound leg of the call. It looks like there is an immediate hangup after the b leg answers. Get a siptrace of that traffic and look to see what is causing the hangup.


-MC


On Fri, May 27, 2011 at 2:27 PM, Yungwei Chen <yungwei at resolvity.com> wrote:

The whole picture looks like the following:
A land-line<-->SIP provider A<-->FS<-->SIP provider A<-->a cell phone

Is there any configuration setting that needs to be enabled or set differently so that those 2 endpoints can talk to each other?
Btw, the same scenario works with Asterisk so my SIP provider shouldn't be the problem.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yungwei Chen
Sent: Friday, May 27, 2011 11:34 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Transferring to a cell phone

Here's the debug log, http://pastebin.freeswitch.org/16398

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Friday, May 27, 2011 10:08 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Transferring to a cell phone

You need to put a debug log on pastebin.freeswitch.org. Be sure to use "FreeSWITCH Log" for the syntax highlighting.
-MC
On Fri, May 27, 2011 at 7:41 AM, Yungwei Chen <yungwei at resolvity.com> wrote:
No.

I just verified that making an outbound call to my cell phone still works.
I even recorded the session just to make sure, and in the recording I hear things both ways.
I first dial 9911 from my SIP client (behind freeswitch), and this leads to the javascript program below.
  session.answer();
  var new_session = new Session("{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");
  session.hangup(16); // disconnects the session between the SIP client and freeswitch
  new_session.execute("record_session", "/tmp/call-from-js-to-cell.wav");
  while (new_session.ready())
  {
      new_session.streamFile("/path/to/local/wav/file");
  }

Now back to the case I'm having problem with.
In this case, I first make a call from a landline to freeswitch through my sip provider, and then a javascript program takes over.
I want to transfer the call to a cell phone so that the landline and the cell phone can communicate with each other.
Here's the javascript program:
  session.answer();

  if (session.ready())
  {
      session.execute("bridge", "{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");
  }

So what am I missing here?

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of mazilo
Sent: Friday, May 27, 2011 7:28 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Transferring to a cell phone


Yungwei Chen wrote:
>
> Thanks for your reply. Using bridge fixed the problem. But I cannot hear
> anything both ways. Any idea?
After switching to using bridge function, does this also happen when you
make an outbound call to your cell phone using your javascript?

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Transferring-to-a-cell-phone-tp6408246p6410887.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

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From dujinfang at gmail.com  Thu Jun  2 06:15:47 2011
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 2 Jun 2011 10:15:47 +0800
Subject: [Freeswitch-users] this gona be fun?
Message-ID: <01E082F916D24B77ACD886187B1EF913@gmail.com>

 http://sites.google.com/site/webrtc/faq

also, does that means iSAC available freely?

-- 
Seven Du
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj: http://www.freeswitch.org.cn
Sent with Sparrow
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From u2nsam at gmail.com  Thu Jun  2 10:48:49 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 2 Jun 2011 12:18:49 +0530
Subject: [Freeswitch-users] manipulate CID
Message-ID: <BANLkTi=ONknfmsocwr+4sQTK54RYUSCTcQ@mail.gmail.com>

Hello,


How do i manipulate callerid number ... i have the below statement

<action application="set"
data="effective_caller_id_number=66${outbound_caller_id_number}"/>

here i want to prefix caller id with 66 so the effective caller id becomes
66 + original callerid .

using above statement the caller id passed was just 66 to the far end.

below is the dialplan i used.

<extension name="7001">
            <condition field="destination_number" expression="^(7001)"
continue="false">
            <action application="set" data="call_timeout=20"/>
            <action application="info" />
            <action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED"/>

            <action application="set"
data="effective_caller_id_number=66${outbound_caller_id_number}"/>
            <action application="ring_ready" />
            <action application="bridge"
data="{sip_cid_type=rpid}sofia/gateway/outbound/1646xxxxxxx"/>

        </extension>



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

This is from header i get .

From: "" <sip:66 at 203.153.53.138>

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



Dialplan: FreeTDM/1:29/7001 Action info()
Dialplan: FreeTDM/1:29/7001 Action
set(continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED)
Dialplan: FreeTDM/1:29/7001 Action
set(effective_caller_id_number=66${outbound_caller_id_number})
Dialplan: FreeTDM/1:29/7001 Action ring_ready()
Dialplan: FreeTDM/1:29/7001 Action
bridge({sip_cid_type=rpid}sofia/gateway/outbound/1646xxxxxxx)
2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:119
(FreeTDM/1:29/7001) State Change CS_ROUTING -> CS_EXECUTE
2011-06-02 06:43:48.111347 [DEBUG] switch_core_session.c:1116 Send signal
FreeTDM/1:29/7001 [BREAK]
2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:359
(FreeTDM/1:29/7001) State ROUTING going to sleep
2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:320
(FreeTDM/1:29/7001) Running State Change CS_EXECUTE
2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:366
(FreeTDM/1:29/7001) State EXECUTE
2011-06-02 06:43:48.111347 [DEBUG] mod_freetdm.c:451 FreeTDM/1:29/7001
CHANNEL EXECUTE
2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:157
FreeTDM/1:29/7001 Standard EXECUTE
EXECUTE FreeTDM/1:29/7001 set(call_timeout=20)
2011-06-02 06:43:48.111347 [DEBUG] mod_dptools.c:1059 FreeTDM/1:29/7001 SET
[call_timeout]=[20]
EXECUTE FreeTDM/1:29/7001 info()
2011-06-02 06:43:48.111347 [INFO] mod_dptools.c:1202 CHANNEL_DATA:
Channel-State: [CS_EXECUTE]
Channel-Call-State: [RINGING]
Channel-State-Number: [4]
Channel-Name: [FreeTDM/1:29/7001]
Unique-ID: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Channel-Call-UUID: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
Answer-State: [ringing]
Channel-Read-Codec-Name: [PCMA]
Channel-Read-Codec-Rate: [8000]
Channel-Read-Codec-Bit-Rate: [64000]
Channel-Write-Codec-Name: [PCMA]
Channel-Write-Codec-Rate: [8000]
Channel-Write-Codec-Bit-Rate: [64000]
Caller-Direction: [inbound]
Caller-Username: [FreeTDM]
Caller-Dialplan: [XML]
Caller-Caller-ID-Number: [8097328707]
Caller-ANI: [8097328707]
Caller-Destination-Number: [7001]
Caller-Unique-ID: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
Caller-Source: [mod_freetdm]
Caller-RDNIS: [7001]
Caller-Channel-Name: [FreeTDM/1:29/7001]
Caller-Profile-Index: [2]
Caller-Profile-Created-Time: [1306997028110372]
Caller-Channel-Created-Time: [1306997028103773]
Caller-Channel-Answered-Time: [0]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
variable_direction: [inbound]
variable_uuid: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
variable_read_codec: [PCMA]
variable_read_rate: [8000]
variable_write_codec: [PCMA]
variable_write_rate: [8000]
variable_channel_name: [FreeTDM/1:29/7001]
variable_freetdm_span_name: [wp1]
variable_freetdm_span_number: [1]
variable_freetdm_chan_number: [29]
variable_freetdm_bearer_capability: [0]
variable_freetdm_bearer_layer1: [3]
variable_freetdm_screening_ind: [network-provided]
variable_freetdm_presentation_ind: [presentation-allowed]
variable_dialed_extension: [7001]
variable_transfer_ringback: [local_stream://moh]
variable_#: [true]
variable_export_vars: [#,*,#]
variable_bind_meta_key: [#]
variable_hangup_after_bridge: [true]
variable_continue_on_fail: [true]
variable_called_party_callgroup: [1]
variable_dialed_user: [7001]
variable_dialed_domain: [192.168.2.190]
variable_call_uuid: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
variable_originate_disposition: [USER_NOT_REGISTERED]
variable_max_forwards: [70]
variable_call_timeout: [20]
variable_current_application: [info]



Regards
Sam
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From avi at avimarcus.net  Thu Jun  2 11:30:22 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 2 Jun 2011 10:30:22 +0300
Subject: [Freeswitch-users] manipulate CID
In-Reply-To: <BANLkTi=ONknfmsocwr+4sQTK54RYUSCTcQ@mail.gmail.com>
References: <BANLkTi=ONknfmsocwr+4sQTK54RYUSCTcQ@mail.gmail.com>
Message-ID: <BANLkTimRhby2uMp83wttpVZb--2wiYHekA@mail.gmail.com>

If you check that info list, ${outbound_caller_id_number} doesn't exist.
It's not a variable programmed into FS by default, although I think the
default dial plan does use it.
Try again with ${effective_caller_id_number}

-Avi

On Thu, Jun 2, 2011 at 9:48 AM, Sam <u2nsam at gmail.com> wrote:

> Hello,
>
>
> How do i manipulate callerid number ... i have the below statement
>
> <action application="set"
> data="effective_caller_id_number=66${outbound_caller_id_number}"/>
>
> here i want to prefix caller id with 66 so the effective caller id becomes
> 66 + original callerid .
>
> using above statement the caller id passed was just 66 to the far end.
>
> below is the dialplan i used.
>
> <extension name="7001">
>             <condition field="destination_number" expression="^(7001)"
> continue="false">
>             <action application="set" data="call_timeout=20"/>
>             <action application="info" />
>             <action application="set"
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED"/>
>
>             <action application="set"
> data="effective_caller_id_number=66${outbound_caller_id_number}"/>
>             <action application="ring_ready" />
>             <action application="bridge"
> data="{sip_cid_type=rpid}sofia/gateway/outbound/1646xxxxxxx"/>
>
>         </extension>
>
>
>
>
> ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
>
> This is from header i get .
>
> From: "" <sip:66 at 203.153.53.138>
>
>
> -------------------------------------------------------------------------------------------------------------------
>
>
>
> Dialplan: FreeTDM/1:29/7001 Action info()
> Dialplan: FreeTDM/1:29/7001 Action
> set(continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED)
> Dialplan: FreeTDM/1:29/7001 Action
> set(effective_caller_id_number=66${outbound_caller_id_number})
> Dialplan: FreeTDM/1:29/7001 Action ring_ready()
> Dialplan: FreeTDM/1:29/7001 Action
> bridge({sip_cid_type=rpid}sofia/gateway/outbound/1646xxxxxxx)
> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:119
> (FreeTDM/1:29/7001) State Change CS_ROUTING -> CS_EXECUTE
> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_session.c:1116 Send signal
> FreeTDM/1:29/7001 [BREAK]
> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:359
> (FreeTDM/1:29/7001) State ROUTING going to sleep
> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:320
> (FreeTDM/1:29/7001) Running State Change CS_EXECUTE
> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:366
> (FreeTDM/1:29/7001) State EXECUTE
> 2011-06-02 06:43:48.111347 [DEBUG] mod_freetdm.c:451 FreeTDM/1:29/7001
> CHANNEL EXECUTE
> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:157
> FreeTDM/1:29/7001 Standard EXECUTE
> EXECUTE FreeTDM/1:29/7001 set(call_timeout=20)
> 2011-06-02 06:43:48.111347 [DEBUG] mod_dptools.c:1059 FreeTDM/1:29/7001 SET
> [call_timeout]=[20]
> EXECUTE FreeTDM/1:29/7001 info()
> 2011-06-02 06:43:48.111347 [INFO] mod_dptools.c:1202 CHANNEL_DATA:
> Channel-State: [CS_EXECUTE]
> Channel-Call-State: [RINGING]
> Channel-State-Number: [4]
> Channel-Name: [FreeTDM/1:29/7001]
> Unique-ID: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
> Call-Direction: [inbound]
> Presence-Call-Direction: [inbound]
> Channel-Call-UUID: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
> Answer-State: [ringing]
> Channel-Read-Codec-Name: [PCMA]
> Channel-Read-Codec-Rate: [8000]
> Channel-Read-Codec-Bit-Rate: [64000]
> Channel-Write-Codec-Name: [PCMA]
> Channel-Write-Codec-Rate: [8000]
> Channel-Write-Codec-Bit-Rate: [64000]
> Caller-Direction: [inbound]
> Caller-Username: [FreeTDM]
> Caller-Dialplan: [XML]
> Caller-Caller-ID-Number: [8097328707]
> Caller-ANI: [8097328707]
> Caller-Destination-Number: [7001]
> Caller-Unique-ID: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
> Caller-Source: [mod_freetdm]
> Caller-RDNIS: [7001]
> Caller-Channel-Name: [FreeTDM/1:29/7001]
> Caller-Profile-Index: [2]
> Caller-Profile-Created-Time: [1306997028110372]
> Caller-Channel-Created-Time: [1306997028103773]
> Caller-Channel-Answered-Time: [0]
> Caller-Channel-Progress-Time: [0]
> Caller-Channel-Progress-Media-Time: [0]
> Caller-Channel-Hangup-Time: [0]
> Caller-Channel-Transfer-Time: [0]
> Caller-Screen-Bit: [true]
> Caller-Privacy-Hide-Name: [false]
> Caller-Privacy-Hide-Number: [false]
> variable_direction: [inbound]
> variable_uuid: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
> variable_read_codec: [PCMA]
> variable_read_rate: [8000]
> variable_write_codec: [PCMA]
> variable_write_rate: [8000]
> variable_channel_name: [FreeTDM/1:29/7001]
> variable_freetdm_span_name: [wp1]
> variable_freetdm_span_number: [1]
> variable_freetdm_chan_number: [29]
> variable_freetdm_bearer_capability: [0]
> variable_freetdm_bearer_layer1: [3]
> variable_freetdm_screening_ind: [network-provided]
> variable_freetdm_presentation_ind: [presentation-allowed]
> variable_dialed_extension: [7001]
> variable_transfer_ringback: [local_stream://moh]
> variable_#: [true]
> variable_export_vars: [#,*,#]
> variable_bind_meta_key: [#]
> variable_hangup_after_bridge: [true]
> variable_continue_on_fail: [true]
> variable_called_party_callgroup: [1]
> variable_dialed_user: [7001]
> variable_dialed_domain: [192.168.2.190]
> variable_call_uuid: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
> variable_originate_disposition: [USER_NOT_REGISTERED]
> variable_max_forwards: [70]
> variable_call_timeout: [20]
> variable_current_application: [info]
>
>
>
> Regards
> Sam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From peter.olsson at visionutveckling.se  Thu Jun  2 13:10:46 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 2 Jun 2011 11:10:46 +0200
Subject: [Freeswitch-users] ClueCon 2011 News and Notes
In-Reply-To: <BANLkTikEUdt69NTcP+gpyJL+x4LJfwBTeA@mail.gmail.com>
References: <00000130481a3868-69cf136f-c857-407d-90c0-e75120e27c52-000000@email.amazonses.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DE5494FE@cooper>,
	<BANLkTikEUdt69NTcP+gpyJL+x4LJfwBTeA@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59DE2B6324@cooper>

Here we go!

Peter Olsson
Email: peter.olsson at visionutveckling.se
Phone: +46 708 389 002

Mikael Norberg
Email: mikael.norberg at visionutveckling.se
Phone: +46 708 389 078

We both work for the company "Visionutveckling AB", you can check our homepage http://www.visionutveckling.se - however, the information in English is quite limited :)

If you want any further information, please contact me.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Michael Collins [msc at freeswitch.org]
Skickat: den 1 juni 2011 22:42
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] ClueCon 2011 News and Notes

Peter,

Can you give me the names and contact information of everyone in your party?

Thanks!
-MC

On Wed, Jun 1, 2011 at 2:06 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
I booked everything today, for me and a collegue ? I?m really looking forward to it!

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] F?r marketing at cluecon.com<mailto:marketing at cluecon.com>
Skickat: den 1 juni 2011 00:09
Till: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
?mne: [Freeswitch-users] ClueCon 2011 News and Notes

ClueCon 2011 Reminder: Extra Chances to Win

We wanted to remind everyone that you still have one more day to book ClueCon 2011 and receive FOUR entries into big giveaways for the coolest laptop and iPad<http://www.cluecon.com/coolest>. (Promotion ends on June 1.) Will you be the big winner? Book now and increase your chances!

For those wanting to know more about what's happening at ClueCon this year we've posted our preliminary schedule<http://www.cluecon.com/schedule>. There are some great talks planned and, of course, a few surprises. Be sure to check back often to see the latest speakers and sponsors.

Visit us at http://www.cluecon.com for more information, or call us at 877.742.CLUE (2583) to inquire about registration, sponsorships, or any other questions you may have. See you in August!

The ClueCon Team
http://www.cluecon.com
877.742.CLUE

[cid:~WRD000.jpg]<http://www.cluecon.com>


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


!DSPAM:4de6a4ed32761769813168!


From u2nsam at gmail.com  Thu Jun  2 13:26:39 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 2 Jun 2011 14:56:39 +0530
Subject: [Freeswitch-users] manipulate CID
In-Reply-To: <BANLkTimRhby2uMp83wttpVZb--2wiYHekA@mail.gmail.com>
References: <BANLkTi=ONknfmsocwr+4sQTK54RYUSCTcQ@mail.gmail.com>
	<BANLkTimRhby2uMp83wttpVZb--2wiYHekA@mail.gmail.com>
Message-ID: <BANLkTin2_v-bYbdPs-=Y9y_-e8gLgxo-hg@mail.gmail.com>

I forgot to mention,

If i use just
<action application="set"
data="effective_caller_id_number=${outbound_caller_id_number}"/>

then the cli is passed but when i use prefix 66 before
${outbound_caller_id_number} the cli is not passed.

my aim is to get the prefix in cli .

Regards
Sam





On Thu, Jun 2, 2011 at 1:00 PM, Avi Marcus <avi at avimarcus.net> wrote:

> If you check that info list, ${outbound_caller_id_number} doesn't exist.
> It's not a variable programmed into FS by default, although I think the
> default dial plan does use it.
> Try again with ${effective_caller_id_number}
>
> -Avi
>
> On Thu, Jun 2, 2011 at 9:48 AM, Sam <u2nsam at gmail.com> wrote:
>
>> Hello,
>>
>>
>> How do i manipulate callerid number ... i have the below statement
>>
>> <action application="set"
>> data="effective_caller_id_number=66${outbound_caller_id_number}"/>
>>
>> here i want to prefix caller id with 66 so the effective caller id becomes
>> 66 + original callerid .
>>
>> using above statement the caller id passed was just 66 to the far end.
>>
>> below is the dialplan i used.
>>
>> <extension name="7001">
>>             <condition field="destination_number" expression="^(7001)"
>> continue="false">
>>             <action application="set" data="call_timeout=20"/>
>>             <action application="info" />
>>             <action application="set"
>> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED"/>
>>
>>             <action application="set"
>> data="effective_caller_id_number=66${outbound_caller_id_number}"/>
>>             <action application="ring_ready" />
>>             <action application="bridge"
>> data="{sip_cid_type=rpid}sofia/gateway/outbound/1646xxxxxxx"/>
>>
>>         </extension>
>>
>>
>>
>>
>> ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
>>
>> This is from header i get .
>>
>> From: "" <sip:66 at 203.153.53.138>
>>
>>
>> -------------------------------------------------------------------------------------------------------------------
>>
>>
>>
>> Dialplan: FreeTDM/1:29/7001 Action info()
>> Dialplan: FreeTDM/1:29/7001 Action
>> set(continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED)
>> Dialplan: FreeTDM/1:29/7001 Action
>> set(effective_caller_id_number=66${outbound_caller_id_number})
>> Dialplan: FreeTDM/1:29/7001 Action ring_ready()
>> Dialplan: FreeTDM/1:29/7001 Action
>> bridge({sip_cid_type=rpid}sofia/gateway/outbound/1646xxxxxxx)
>> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:119
>> (FreeTDM/1:29/7001) State Change CS_ROUTING -> CS_EXECUTE
>> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_session.c:1116 Send signal
>> FreeTDM/1:29/7001 [BREAK]
>> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:359
>> (FreeTDM/1:29/7001) State ROUTING going to sleep
>> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:320
>> (FreeTDM/1:29/7001) Running State Change CS_EXECUTE
>> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:366
>> (FreeTDM/1:29/7001) State EXECUTE
>> 2011-06-02 06:43:48.111347 [DEBUG] mod_freetdm.c:451 FreeTDM/1:29/7001
>> CHANNEL EXECUTE
>> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:157
>> FreeTDM/1:29/7001 Standard EXECUTE
>> EXECUTE FreeTDM/1:29/7001 set(call_timeout=20)
>> 2011-06-02 06:43:48.111347 [DEBUG] mod_dptools.c:1059 FreeTDM/1:29/7001
>> SET [call_timeout]=[20]
>> EXECUTE FreeTDM/1:29/7001 info()
>> 2011-06-02 06:43:48.111347 [INFO] mod_dptools.c:1202 CHANNEL_DATA:
>> Channel-State: [CS_EXECUTE]
>> Channel-Call-State: [RINGING]
>> Channel-State-Number: [4]
>> Channel-Name: [FreeTDM/1:29/7001]
>> Unique-ID: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
>> Call-Direction: [inbound]
>> Presence-Call-Direction: [inbound]
>> Channel-Call-UUID: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
>> Answer-State: [ringing]
>> Channel-Read-Codec-Name: [PCMA]
>> Channel-Read-Codec-Rate: [8000]
>> Channel-Read-Codec-Bit-Rate: [64000]
>> Channel-Write-Codec-Name: [PCMA]
>> Channel-Write-Codec-Rate: [8000]
>> Channel-Write-Codec-Bit-Rate: [64000]
>> Caller-Direction: [inbound]
>> Caller-Username: [FreeTDM]
>> Caller-Dialplan: [XML]
>> Caller-Caller-ID-Number: [8097328707]
>> Caller-ANI: [8097328707]
>> Caller-Destination-Number: [7001]
>> Caller-Unique-ID: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
>> Caller-Source: [mod_freetdm]
>> Caller-RDNIS: [7001]
>> Caller-Channel-Name: [FreeTDM/1:29/7001]
>> Caller-Profile-Index: [2]
>> Caller-Profile-Created-Time: [1306997028110372]
>> Caller-Channel-Created-Time: [1306997028103773]
>> Caller-Channel-Answered-Time: [0]
>> Caller-Channel-Progress-Time: [0]
>> Caller-Channel-Progress-Media-Time: [0]
>> Caller-Channel-Hangup-Time: [0]
>> Caller-Channel-Transfer-Time: [0]
>> Caller-Screen-Bit: [true]
>> Caller-Privacy-Hide-Name: [false]
>> Caller-Privacy-Hide-Number: [false]
>> variable_direction: [inbound]
>> variable_uuid: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
>> variable_read_codec: [PCMA]
>> variable_read_rate: [8000]
>> variable_write_codec: [PCMA]
>> variable_write_rate: [8000]
>> variable_channel_name: [FreeTDM/1:29/7001]
>> variable_freetdm_span_name: [wp1]
>> variable_freetdm_span_number: [1]
>> variable_freetdm_chan_number: [29]
>> variable_freetdm_bearer_capability: [0]
>> variable_freetdm_bearer_layer1: [3]
>> variable_freetdm_screening_ind: [network-provided]
>> variable_freetdm_presentation_ind: [presentation-allowed]
>> variable_dialed_extension: [7001]
>> variable_transfer_ringback: [local_stream://moh]
>> variable_#: [true]
>> variable_export_vars: [#,*,#]
>> variable_bind_meta_key: [#]
>> variable_hangup_after_bridge: [true]
>> variable_continue_on_fail: [true]
>> variable_called_party_callgroup: [1]
>> variable_dialed_user: [7001]
>> variable_dialed_domain: [192.168.2.190]
>> variable_call_uuid: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
>> variable_originate_disposition: [USER_NOT_REGISTERED]
>> variable_max_forwards: [70]
>> variable_call_timeout: [20]
>> variable_current_application: [info]
>>
>>
>>
>> Regards
>> Sam
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jan.berger at video24.no  Thu Jun  2 14:25:34 2011
From: jan.berger at video24.no (Jan Berger)
Date: Thu, 2 Jun 2011 12:25:34 +0200
Subject: [Freeswitch-users] mrcp
Message-ID: <A358176020C045358C04C08944501FAA@dell9400>

Hi,

 

I would like to communicate with MRCP to control TTS and ASR from a C module
- how can I do that easiest?

 

Jan

 

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From ovvenkatesan at gmail.com  Thu Jun  2 14:33:05 2011
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Thu, 2 Jun 2011 16:03:05 +0530
Subject: [Freeswitch-users] Freeswitch Server getting down
Message-ID: <BANLkTineWK6wstnKzdbM=rC31D9OBOfwZA@mail.gmail.com>

Hi to all,

I dont know whether this problem related to PRI line (OR) wanpipe driver (
OR ) PRI Card

I use Linux fedora 13  and Sangoma A101 PRI  card to handle incoming calls
from E1 lines.

Everything works fine for some time.

After that, I could see  cli debug output saying,

all the channels are disconnected and then immediately
it says that "signaling status changed to UP".

After some time, freeswitch getting shutdown saying
ftmod_wanpipe.c:967 [s1c31][1:16] Failed to read from sangoma device: No
buffer space available (-65)


I pastebin debug logs for the same.

http://pastebin.freeswitch.org/16432

I am failed to identify the problem.
Can you anyone guide me what is going wrong?



Regards
Venkatesan OV.
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From avi at avimarcus.net  Thu Jun  2 15:12:08 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 2 Jun 2011 14:12:08 +0300
Subject: [Freeswitch-users] manipulate CID
In-Reply-To: <BANLkTin2_v-bYbdPs-=Y9y_-e8gLgxo-hg@mail.gmail.com>
References: <BANLkTi=ONknfmsocwr+4sQTK54RYUSCTcQ@mail.gmail.com>
	<BANLkTimRhby2uMp83wttpVZb--2wiYHekA@mail.gmail.com>
	<BANLkTin2_v-bYbdPs-=Y9y_-e8gLgxo-hg@mail.gmail.com>
Message-ID: <BANLkTikUGAconHy5mG1A1j_D4W3J4tuKDg@mail.gmail.com>

In the exact same part of the dialplan? If so, weird.
Can you pastebin a trace on /log 7 from the fs_cli with the xml this way,
and then prefix 66 to the xml, and do a second trace?

-Avi


On Thu, Jun 2, 2011 at 12:26 PM, Sam <u2nsam at gmail.com> wrote:

> I forgot to mention,
>
> If i use just
> <action application="set"
> data="effective_caller_id_number=${outbound_caller_id_number}"/>
>
> then the cli is passed but when i use prefix 66 before
> ${outbound_caller_id_number} the cli is not passed.
>
> my aim is to get the prefix in cli .
>
> Regards
> Sam
>
>
>
>
>
>
> On Thu, Jun 2, 2011 at 1:00 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> If you check that info list, ${outbound_caller_id_number} doesn't exist.
>> It's not a variable programmed into FS by default, although I think the
>> default dial plan does use it.
>> Try again with ${effective_caller_id_number}
>>
>> -Avi
>>
>> On Thu, Jun 2, 2011 at 9:48 AM, Sam <u2nsam at gmail.com> wrote:
>>
>>> Hello,
>>>
>>>
>>> How do i manipulate callerid number ... i have the below statement
>>>
>>> <action application="set"
>>> data="effective_caller_id_number=66${outbound_caller_id_number}"/>
>>>
>>> here i want to prefix caller id with 66 so the effective caller id
>>> becomes 66 + original callerid .
>>>
>>> using above statement the caller id passed was just 66 to the far end.
>>>
>>> below is the dialplan i used.
>>>
>>> <extension name="7001">
>>>             <condition field="destination_number" expression="^(7001)"
>>> continue="false">
>>>             <action application="set" data="call_timeout=20"/>
>>>             <action application="info" />
>>>             <action application="set"
>>> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED"/>
>>>
>>>             <action application="set"
>>> data="effective_caller_id_number=66${outbound_caller_id_number}"/>
>>>             <action application="ring_ready" />
>>>             <action application="bridge"
>>> data="{sip_cid_type=rpid}sofia/gateway/outbound/1646xxxxxxx"/>
>>>
>>>         </extension>
>>>
>>>
>>>
>>>
>>> ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
>>>
>>> This is from header i get .
>>>
>>> From: "" <sip:66 at 203.153.53.138>
>>>
>>>
>>> -------------------------------------------------------------------------------------------------------------------
>>>
>>>
>>>
>>> Dialplan: FreeTDM/1:29/7001 Action info()
>>> Dialplan: FreeTDM/1:29/7001 Action
>>> set(continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED)
>>> Dialplan: FreeTDM/1:29/7001 Action
>>> set(effective_caller_id_number=66${outbound_caller_id_number})
>>> Dialplan: FreeTDM/1:29/7001 Action ring_ready()
>>> Dialplan: FreeTDM/1:29/7001 Action
>>> bridge({sip_cid_type=rpid}sofia/gateway/outbound/1646xxxxxxx)
>>> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:119
>>> (FreeTDM/1:29/7001) State Change CS_ROUTING -> CS_EXECUTE
>>> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_session.c:1116 Send signal
>>> FreeTDM/1:29/7001 [BREAK]
>>> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:359
>>> (FreeTDM/1:29/7001) State ROUTING going to sleep
>>> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:320
>>> (FreeTDM/1:29/7001) Running State Change CS_EXECUTE
>>> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:366
>>> (FreeTDM/1:29/7001) State EXECUTE
>>> 2011-06-02 06:43:48.111347 [DEBUG] mod_freetdm.c:451 FreeTDM/1:29/7001
>>> CHANNEL EXECUTE
>>> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:157
>>> FreeTDM/1:29/7001 Standard EXECUTE
>>> EXECUTE FreeTDM/1:29/7001 set(call_timeout=20)
>>> 2011-06-02 06:43:48.111347 [DEBUG] mod_dptools.c:1059 FreeTDM/1:29/7001
>>> SET [call_timeout]=[20]
>>> EXECUTE FreeTDM/1:29/7001 info()
>>> 2011-06-02 06:43:48.111347 [INFO] mod_dptools.c:1202 CHANNEL_DATA:
>>> Channel-State: [CS_EXECUTE]
>>> Channel-Call-State: [RINGING]
>>> Channel-State-Number: [4]
>>> Channel-Name: [FreeTDM/1:29/7001]
>>> Unique-ID: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
>>> Call-Direction: [inbound]
>>> Presence-Call-Direction: [inbound]
>>> Channel-Call-UUID: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
>>> Answer-State: [ringing]
>>> Channel-Read-Codec-Name: [PCMA]
>>> Channel-Read-Codec-Rate: [8000]
>>> Channel-Read-Codec-Bit-Rate: [64000]
>>> Channel-Write-Codec-Name: [PCMA]
>>> Channel-Write-Codec-Rate: [8000]
>>> Channel-Write-Codec-Bit-Rate: [64000]
>>> Caller-Direction: [inbound]
>>> Caller-Username: [FreeTDM]
>>> Caller-Dialplan: [XML]
>>> Caller-Caller-ID-Number: [8097328707]
>>> Caller-ANI: [8097328707]
>>> Caller-Destination-Number: [7001]
>>> Caller-Unique-ID: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
>>> Caller-Source: [mod_freetdm]
>>> Caller-RDNIS: [7001]
>>> Caller-Channel-Name: [FreeTDM/1:29/7001]
>>> Caller-Profile-Index: [2]
>>> Caller-Profile-Created-Time: [1306997028110372]
>>> Caller-Channel-Created-Time: [1306997028103773]
>>> Caller-Channel-Answered-Time: [0]
>>> Caller-Channel-Progress-Time: [0]
>>> Caller-Channel-Progress-Media-Time: [0]
>>> Caller-Channel-Hangup-Time: [0]
>>> Caller-Channel-Transfer-Time: [0]
>>> Caller-Screen-Bit: [true]
>>> Caller-Privacy-Hide-Name: [false]
>>> Caller-Privacy-Hide-Number: [false]
>>> variable_direction: [inbound]
>>> variable_uuid: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
>>> variable_read_codec: [PCMA]
>>> variable_read_rate: [8000]
>>> variable_write_codec: [PCMA]
>>> variable_write_rate: [8000]
>>> variable_channel_name: [FreeTDM/1:29/7001]
>>> variable_freetdm_span_name: [wp1]
>>> variable_freetdm_span_number: [1]
>>> variable_freetdm_chan_number: [29]
>>> variable_freetdm_bearer_capability: [0]
>>> variable_freetdm_bearer_layer1: [3]
>>> variable_freetdm_screening_ind: [network-provided]
>>> variable_freetdm_presentation_ind: [presentation-allowed]
>>> variable_dialed_extension: [7001]
>>> variable_transfer_ringback: [local_stream://moh]
>>> variable_#: [true]
>>> variable_export_vars: [#,*,#]
>>> variable_bind_meta_key: [#]
>>> variable_hangup_after_bridge: [true]
>>> variable_continue_on_fail: [true]
>>> variable_called_party_callgroup: [1]
>>> variable_dialed_user: [7001]
>>> variable_dialed_domain: [192.168.2.190]
>>> variable_call_uuid: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
>>> variable_originate_disposition: [USER_NOT_REGISTERED]
>>> variable_max_forwards: [70]
>>> variable_call_timeout: [20]
>>> variable_current_application: [info]
>>>
>>>
>>>
>>> Regards
>>> Sam
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Thu Jun  2 15:43:38 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 2 Jun 2011 12:43:38 +0100
Subject: [Freeswitch-users] mrcp
In-Reply-To: <A358176020C045358C04C08944501FAA@dell9400>
References: <A358176020C045358C04C08944501FAA@dell9400>
Message-ID: <BANLkTi=Q6oMRUNq9GSVjMORwxS0eWV2=9w@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Mod_unimrcp

:)



On 2 June 2011 11:25, Jan Berger <jan.berger at video24.no> wrote:

>  Hi,
>
>
>
> I would like to communicate with MRCP to control TTS and ASR from a C
> module - how can I do that easiest?
>
>
>
> Jan
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From freeswitch at aastral.net  Thu Jun  2 17:29:57 2011
From: freeswitch at aastral.net (Bill W.)
Date: Thu, 02 Jun 2011 09:29:57 -0400
Subject: [Freeswitch-users] intermittent EXCHANGE_ROUTING_ERROR
In-Reply-To: <BANLkTikXJrxuwUXAd2kYFCiDQOLGQgw9dg@mail.gmail.com>
References: <1QRrkF-0007cj-IK@mail.aastral.net>	<BANLkTikc5bMokDY6__miju=rOUmsLc7kNg@mail.gmail.com>	<1QRt7M-0002Zm-GE@mail.aastral.net>
	<BANLkTikXJrxuwUXAd2kYFCiDQOLGQgw9dg@mail.gmail.com>
Message-ID: <1QS7yM-0005cW-8h@mail.aastral.net>

Hey Michael,

Thanks so much for the advice.  I've already enabled uuid logging, and
have grepped the log for the uuid of the call that failed.  As in my
initial email, it just goes from 'setting variables' to the error.  No
new channel is created.

There are no transfers in the dialplan nor in freeswitch.log.

I don't have XML CDRs enabled, but the cdr-csv  log entry is unremarkable.

I'll enable XML CDR and see if that can provide some more information.


On 6/1/11 5:46 PM, Michael Collins wrote:
> Well, if it's inside the dialplan you'll see it in the log. You may want
> to rotate logs frequently while tracking this down. It might also help
> if you enable uuid logging so that you can filter down to just a single
> call's log lines. You may also wish to examine the XML CDRs for these
> calls. I'd be curious to see what's in the call flows.
> 
> -MC
> 
> On Wed, Jun 1, 2011 at 2:38 PM, Bill W. <freeswitch at aastral.net
> <mailto:freeswitch at aastral.net>> wrote:
> 
>     Hey Anthony,
> 
>     Thanks very much for your input.  Can you recommend a way I can prove
>     that to myself?  Is there some sort of logging to turn on to see that?
> 
>     The reason I ask is that for this instance I receive a call in on one
>     sofia profile and then use the bridge command to send it out another
>     sofia profile. I don't use the transfer application to complete the
>     call.
> 
> 
> 
>     On 6/1/11 4:36 PM, Anthony Minessale wrote:
>     > routing loop, its transferring more that 73 times inside the dialplan
>     >
>     > On Wed, Jun 1, 2011 at 3:10 PM, Bill W. <freeswitch at aastral.net
>     <mailto:freeswitch at aastral.net>> wrote:
>     >> Hey All,
>     >>
>     >> About once or twice a day, I'm getting an EXCHANGE_ROUTING_ERROR
>     failure.
>     >>
>     >> I've enabled debugging, including logging SIP messages, and can't
>     seem
>     >> to find the cause of the error.
>     >>
>     >>
>     >> Freeswitch goes from setting variables to exchange routing error:
>     >> switch_channel.c:935 EXPORTING[export_vars]
>     >> to
>     >> switch_ivr_originate.c:2447 Cannot create outgoing channel of type
>     >> [sofia] cause: [EXCHANGE_ROUTING_ERROR]
>     >>
>     >>
>     >> Whenever that happens, "switch_channel.c:816 New Channel" is never
>     >> called and I don't see any outbound SIP messages to the IP in
>     question.
>     >>
>     >> So if no new channel is created, how can there be a routing error?
>     >>
>     >> Any insight or advice on how to debug this would be greatly
>     appreciated.
>     >>
>     >> Thanks!
>     >>
>     >>
>     >>
>     >> _______________________________________________
>     >> FreeSWITCH-users mailing list
>     >> FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >> http://www.freeswitch.org
>     >>
>     >
>     >
>     >
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From anthony.minessale at gmail.com  Thu Jun  2 18:02:58 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 2 Jun 2011 09:02:58 -0500
Subject: [Freeswitch-users] intermittent EXCHANGE_ROUTING_ERROR
In-Reply-To: <1QS7yM-0005cW-8h@mail.aastral.net>
References: <1QRrkF-0007cj-IK@mail.aastral.net>
	<BANLkTikc5bMokDY6__miju=rOUmsLc7kNg@mail.gmail.com>
	<1QRt7M-0002Zm-GE@mail.aastral.net>
	<BANLkTikXJrxuwUXAd2kYFCiDQOLGQgw9dg@mail.gmail.com>
	<1QS7yM-0005cW-8h@mail.aastral.net>
Message-ID: <BANLkTi=hv5KZhMUfj1TKbucARH8gzADqpw@mail.gmail.com>

its not only transfers, its possibly a call looping in a circle to the same box.

Are you using your own domain in a bridge app somewhere for instance:

<action application="bridge" data="sofia/internal/1234 at domain.com"/>

And if your own box is domain.com and you mistakenly are trying to
call the extension registered to 1234 you would actually call yourself
over sip in a loop up to 73 times until it hungup with that cause.


On Thu, Jun 2, 2011 at 8:29 AM, Bill W. <freeswitch at aastral.net> wrote:
> Hey Michael,
>
> Thanks so much for the advice. ?I've already enabled uuid logging, and
> have grepped the log for the uuid of the call that failed. ?As in my
> initial email, it just goes from 'setting variables' to the error. ?No
> new channel is created.
>
> There are no transfers in the dialplan nor in freeswitch.log.
>
> I don't have XML CDRs enabled, but the cdr-csv ?log entry is unremarkable.
>
> I'll enable XML CDR and see if that can provide some more information.
>
>
> On 6/1/11 5:46 PM, Michael Collins wrote:
>> Well, if it's inside the dialplan you'll see it in the log. You may want
>> to rotate logs frequently while tracking this down. It might also help
>> if you enable uuid logging so that you can filter down to just a single
>> call's log lines. You may also wish to examine the XML CDRs for these
>> calls. I'd be curious to see what's in the call flows.
>>
>> -MC
>>
>> On Wed, Jun 1, 2011 at 2:38 PM, Bill W. <freeswitch at aastral.net
>> <mailto:freeswitch at aastral.net>> wrote:
>>
>> ? ? Hey Anthony,
>>
>> ? ? Thanks very much for your input. ?Can you recommend a way I can prove
>> ? ? that to myself? ?Is there some sort of logging to turn on to see that?
>>
>> ? ? The reason I ask is that for this instance I receive a call in on one
>> ? ? sofia profile and then use the bridge command to send it out another
>> ? ? sofia profile. I don't use the transfer application to complete the
>> ? ? call.
>>
>>
>>
>> ? ? On 6/1/11 4:36 PM, Anthony Minessale wrote:
>> ? ? > routing loop, its transferring more that 73 times inside the dialplan
>> ? ? >
>> ? ? > On Wed, Jun 1, 2011 at 3:10 PM, Bill W. <freeswitch at aastral.net
>> ? ? <mailto:freeswitch at aastral.net>> wrote:
>> ? ? >> Hey All,
>> ? ? >>
>> ? ? >> About once or twice a day, I'm getting an EXCHANGE_ROUTING_ERROR
>> ? ? failure.
>> ? ? >>
>> ? ? >> I've enabled debugging, including logging SIP messages, and can't
>> ? ? seem
>> ? ? >> to find the cause of the error.
>> ? ? >>
>> ? ? >>
>> ? ? >> Freeswitch goes from setting variables to exchange routing error:
>> ? ? >> switch_channel.c:935 EXPORTING[export_vars]
>> ? ? >> to
>> ? ? >> switch_ivr_originate.c:2447 Cannot create outgoing channel of type
>> ? ? >> [sofia] cause: [EXCHANGE_ROUTING_ERROR]
>> ? ? >>
>> ? ? >>
>> ? ? >> Whenever that happens, "switch_channel.c:816 New Channel" is never
>> ? ? >> called and I don't see any outbound SIP messages to the IP in
>> ? ? question.
>> ? ? >>
>> ? ? >> So if no new channel is created, how can there be a routing error?
>> ? ? >>
>> ? ? >> Any insight or advice on how to debug this would be greatly
>> ? ? appreciated.
>> ? ? >>
>> ? ? >> Thanks!
>> ? ? >>
>> ? ? >>
>> ? ? >>
>> ? ? >> _______________________________________________
>> ? ? >> FreeSWITCH-users mailing list
>> ? ? >> FreeSWITCH-users at lists.freeswitch.org
>> ? ? <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> ? ? >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> ? ? >>
>> ? ? UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> ? ? >> http://www.freeswitch.org
>> ? ? >>
>> ? ? >
>> ? ? >
>> ? ? >
>>
>> ? ? _______________________________________________
>> ? ? FreeSWITCH-users mailing list
>> ? ? FreeSWITCH-users at lists.freeswitch.org
>> ? ? <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> ? ? UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> ? ? http://www.freeswitch.org
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From yungwei at resolvity.com  Thu Jun  2 18:34:56 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Thu, 2 Jun 2011 10:34:56 -0400
Subject: [Freeswitch-users] Unable to get events from freeswitch using
 java ESL client
Message-ID: <33095823FD21DF429B481B5163264B7950AC4FB5F9@VMBX102.ihostexchange.net>


I missed the following:
client.setEventSubscriptions("plain", "ALL");


From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] 
Sent: Wednesday, June 01, 2011 6:28 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Unable to get events from freeswitch using java ESL client


Hi,

I am using org.freeswitch.esl.client-0.9.2 to ask freeswitch to make an outbound call in the following java class.
I want to be able to get notified when a call is not answered for whatever reason.
However, I don't get any events from freeswitch. What am I missing here?
Thanks.

public class esl implements IEslEventListener {
 
 /**
  * @param args
  */
 public static void main(String[] args) {
  esl e = new esl();
  Client client = new Client();
  EslMessage response = null;
  
  try {
   client.connect("192.168.6.18", 8021, "ClueCon", 10);
   client.addEventListener(e);
   
   response = client.sendSyncApiCommand("originate", "{ignore_early_media=true}sofia/gateway/broadvoice/1112223333 9886");
   
   CommandResponse cr = client.close();
   System.out.println("Done.");
   
  } catch (InboundConnectionFailure ex) {
   ex.printStackTrace();
  }
 }
 @Override
 public void backgroundJobResultReceived(EslEvent arg0) {
  System.out.println("backgroundJobResultReceived");
 }
 @Override
 public void eventReceived(EslEvent arg0) {
  System.out.println(String.format("eventReceived, event name=%s", arg0.getEventName()));
 }
}

From Nabble at slickdeals.endjunk.com  Thu Jun  2 18:39:02 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 2 Jun 2011 07:39:02 -0700 (PDT)
Subject: [Freeswitch-users] how to use mod_mp4
In-Reply-To: <BANLkTimLwBeTnNi13zNhZnH9KU4uLWTPwA@mail.gmail.com>
References: <1306486049569-6410226.post@n2.nabble.com>
	<BANLkTi=4ky6Kj9gE94k_i7tDvdW+LcpjvQ@mail.gmail.com>
	<1306841276439-6422324.post@n2.nabble.com>
	<BANLkTi=O5A_CfCO6PVeyz+RGOCA6Zv1nug@mail.gmail.com>
	<1306858803865-6423457.post@n2.nabble.com>
	<BANLkTinUkgF1semv1VF1edY0WGBsKhJ67Q@mail.gmail.com>
	<1306927048156-6426476.post@n2.nabble.com>
	<BANLkTimLwBeTnNi13zNhZnH9KU4uLWTPwA@mail.gmail.com>
Message-ID: <1307025542904-6431457.post@n2.nabble.com>


Steven Ayre wrote:
> 
> Should do, it's the same library. :)
I would think so. However, upon a closer look on both 
http://mp4v2.googlecode.com/files/mp4v2-1.9.1.tar.bz2 MP4v2  and 
http://libmp4.svn.sourceforge.net/viewvc/libmp4/trunk/libmp4/ libmp4 
source, it appears MP4v2 uses some different header file names to produce
libmp4v2.so library while libmp4 produces libmp4_static.a library. Perhaps,
those who compiled mod_mp4 on a Linux machine already has libmp4 patched to
produce libmp4v2.so library.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/how-to-use-mod-mp4-tp6410226p6431457.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From brian at freeswitch.org  Thu Jun  2 18:56:55 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 2 Jun 2011 09:56:55 -0500
Subject: [Freeswitch-users] mod_msn
In-Reply-To: <931F80EC287F4BBEA4174FFBE344D831@e1705>
References: <931F80EC287F4BBEA4174FFBE344D831@e1705>
Message-ID: <E5F9A9ED-538C-4BF9-A46A-56DD0C574BB9@freeswitch.org>

Isn't one already in contrib?

/b

On Jun 1, 2011, at 1:23 AM, Madovsky wrote:

> Do you think it's technically feasible to
> make a module that connect to MSN network ?_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Thu Jun  2 18:58:12 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 2 Jun 2011 09:58:12 -0500
Subject: [Freeswitch-users] Forcing SRTP/TLS connections
In-Reply-To: <BANLkTi=feKRsH4gDyaTHNY=3-spnrJtyHw@mail.gmail.com>
References: <BANLkTi==jP5Lpy6p0GdwWT7-SQ7gZcEk5w@mail.gmail.com>
	<BANLkTi=feKRsH4gDyaTHNY=3-spnrJtyHw@mail.gmail.com>
Message-ID: <CD7B7384-9236-4AD5-A0E0-86089E3E751A@freeswitch.org>

What was so wrong with adding ;transport=tls to the URI :P

/b

On May 30, 2011, at 1:05 PM, Mitch Capper wrote:

> You may want to check out my tls patch at:
> 
> it features the option for tls_only forcing only the encrypted port to
> be open: http://jira.freeswitch.org/browse/FS-3071  in your dialplan
> for calls you can set sip_secure_media to true to force it to SRTP.
> 
> ~Mitch

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From mitch.capper at gmail.com  Thu Jun  2 19:39:34 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Thu, 2 Jun 2011 08:39:34 -0700
Subject: [Freeswitch-users] Forcing SRTP/TLS connections
In-Reply-To: <CD7B7384-9236-4AD5-A0E0-86089E3E751A@freeswitch.org>
References: <BANLkTi==jP5Lpy6p0GdwWT7-SQ7gZcEk5w@mail.gmail.com>
	<BANLkTi=feKRsH4gDyaTHNY=3-spnrJtyHw@mail.gmail.com>
	<CD7B7384-9236-4AD5-A0E0-86089E3E751A@freeswitch.org>
Message-ID: <BANLkTi=DqMA2GipVjbMtMa3mE_zaKsyFWg@mail.gmail.com>

This just prevents any unencrypted registration attempts, useful for
also stopping brute force attacks and ensuring clients are properly
configured and not attempting to connect over an insecure channel.

~mitch

On Thu, Jun 2, 2011 at 7:58 AM, Brian West <brian at freeswitch.org> wrote:
> What was so wrong with adding ;transport=tls to the URI :P
> /b
> On May 30, 2011, at 1:05 PM, Mitch Capper wrote:
>
> You may want to check out my tls patch at:
>
> it features the option for tls_only forcing only the encrypted port to
> be open:?http://jira.freeswitch.org/browse/FS-3071??in your dialplan
> for calls you can set sip_secure_media to true to force it to SRTP.
>
> ~Mitch
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From freeswitch at aastral.net  Thu Jun  2 20:07:00 2011
From: freeswitch at aastral.net (Bill W.)
Date: Thu, 02 Jun 2011 12:07:00 -0400
Subject: [Freeswitch-users] intermittent EXCHANGE_ROUTING_ERROR
In-Reply-To: <BANLkTi=hv5KZhMUfj1TKbucARH8gzADqpw@mail.gmail.com>
References: <1QRrkF-0007cj-IK@mail.aastral.net>	<BANLkTikc5bMokDY6__miju=rOUmsLc7kNg@mail.gmail.com>	<1QRt7M-0002Zm-GE@mail.aastral.net>	<BANLkTikXJrxuwUXAd2kYFCiDQOLGQgw9dg@mail.gmail.com>	<1QS7yM-0005cW-8h@mail.aastral.net>
	<BANLkTi=hv5KZhMUfj1TKbucARH8gzADqpw@mail.gmail.com>
Message-ID: <1QSAQM-0003GJ-6U@mail.aastral.net>


Hey Anthony,

All the bridge commands are of the form:
sofia/profile/18005551212 at 1.2.3.4

Also, if it was looping, would I see that in the logs?  It would parse
the dialplan 73 times, correct?  Because I don't see that.






From u2nsam at gmail.com  Thu Jun  2 20:16:45 2011
From: u2nsam at gmail.com (Sam)
Date: Thu, 2 Jun 2011 21:46:45 +0530
Subject: [Freeswitch-users] manipulate CID
In-Reply-To: <BANLkTi=ONknfmsocwr+4sQTK54RYUSCTcQ@mail.gmail.com>
References: <BANLkTi=ONknfmsocwr+4sQTK54RYUSCTcQ@mail.gmail.com>
Message-ID: <BANLkTimWyQ4mzKAWiumtUjwdA4mcw66zUQ@mail.gmail.com>

Is it possible to add prefix to caller id on FS ?

Regds
Sam



On Thu, Jun 2, 2011 at 12:18 PM, Sam <u2nsam at gmail.com> wrote:

> Hello,
>
>
> How do i manipulate callerid number ... i have the below statement
>
> <action application="set"
> data="effective_caller_id_number=66${outbound_caller_id_number}"/>
>
> here i want to prefix caller id with 66 so the effective caller id becomes
> 66 + original callerid .
>
> using above statement the caller id passed was just 66 to the far end.
>
> below is the dialplan i used.
>
> <extension name="7001">
>             <condition field="destination_number" expression="^(7001)"
> continue="false">
>             <action application="set" data="call_timeout=20"/>
>             <action application="info" />
>             <action application="set"
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED"/>
>
>             <action application="set"
> data="effective_caller_id_number=66${outbound_caller_id_number}"/>
>             <action application="ring_ready" />
>             <action application="bridge"
> data="{sip_cid_type=rpid}sofia/gateway/outbound/1646xxxxxxx"/>
>
>         </extension>
>
>
>
>
> ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
>
> This is from header i get .
>
> From: "" <sip:66 at 203.153.53.138>
>
>
> -------------------------------------------------------------------------------------------------------------------
>
>
>
> Dialplan: FreeTDM/1:29/7001 Action info()
> Dialplan: FreeTDM/1:29/7001 Action
> set(continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED)
> Dialplan: FreeTDM/1:29/7001 Action
> set(effective_caller_id_number=66${outbound_caller_id_number})
> Dialplan: FreeTDM/1:29/7001 Action ring_ready()
> Dialplan: FreeTDM/1:29/7001 Action
> bridge({sip_cid_type=rpid}sofia/gateway/outbound/1646xxxxxxx)
> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:119
> (FreeTDM/1:29/7001) State Change CS_ROUTING -> CS_EXECUTE
> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_session.c:1116 Send signal
> FreeTDM/1:29/7001 [BREAK]
> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:359
> (FreeTDM/1:29/7001) State ROUTING going to sleep
> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:320
> (FreeTDM/1:29/7001) Running State Change CS_EXECUTE
> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:366
> (FreeTDM/1:29/7001) State EXECUTE
> 2011-06-02 06:43:48.111347 [DEBUG] mod_freetdm.c:451 FreeTDM/1:29/7001
> CHANNEL EXECUTE
> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:157
> FreeTDM/1:29/7001 Standard EXECUTE
> EXECUTE FreeTDM/1:29/7001 set(call_timeout=20)
> 2011-06-02 06:43:48.111347 [DEBUG] mod_dptools.c:1059 FreeTDM/1:29/7001 SET
> [call_timeout]=[20]
> EXECUTE FreeTDM/1:29/7001 info()
> 2011-06-02 06:43:48.111347 [INFO] mod_dptools.c:1202 CHANNEL_DATA:
> Channel-State: [CS_EXECUTE]
> Channel-Call-State: [RINGING]
> Channel-State-Number: [4]
> Channel-Name: [FreeTDM/1:29/7001]
> Unique-ID: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
> Call-Direction: [inbound]
> Presence-Call-Direction: [inbound]
> Channel-Call-UUID: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
> Answer-State: [ringing]
> Channel-Read-Codec-Name: [PCMA]
> Channel-Read-Codec-Rate: [8000]
> Channel-Read-Codec-Bit-Rate: [64000]
> Channel-Write-Codec-Name: [PCMA]
> Channel-Write-Codec-Rate: [8000]
> Channel-Write-Codec-Bit-Rate: [64000]
> Caller-Direction: [inbound]
> Caller-Username: [FreeTDM]
> Caller-Dialplan: [XML]
> Caller-Caller-ID-Number: [8097328707]
> Caller-ANI: [8097328707]
> Caller-Destination-Number: [7001]
> Caller-Unique-ID: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
> Caller-Source: [mod_freetdm]
> Caller-RDNIS: [7001]
> Caller-Channel-Name: [FreeTDM/1:29/7001]
> Caller-Profile-Index: [2]
> Caller-Profile-Created-Time: [1306997028110372]
> Caller-Channel-Created-Time: [1306997028103773]
> Caller-Channel-Answered-Time: [0]
> Caller-Channel-Progress-Time: [0]
> Caller-Channel-Progress-Media-Time: [0]
> Caller-Channel-Hangup-Time: [0]
> Caller-Channel-Transfer-Time: [0]
> Caller-Screen-Bit: [true]
> Caller-Privacy-Hide-Name: [false]
> Caller-Privacy-Hide-Number: [false]
> variable_direction: [inbound]
> variable_uuid: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
> variable_read_codec: [PCMA]
> variable_read_rate: [8000]
> variable_write_codec: [PCMA]
> variable_write_rate: [8000]
> variable_channel_name: [FreeTDM/1:29/7001]
> variable_freetdm_span_name: [wp1]
> variable_freetdm_span_number: [1]
> variable_freetdm_chan_number: [29]
> variable_freetdm_bearer_capability: [0]
> variable_freetdm_bearer_layer1: [3]
> variable_freetdm_screening_ind: [network-provided]
> variable_freetdm_presentation_ind: [presentation-allowed]
> variable_dialed_extension: [7001]
> variable_transfer_ringback: [local_stream://moh]
> variable_#: [true]
> variable_export_vars: [#,*,#]
> variable_bind_meta_key: [#]
> variable_hangup_after_bridge: [true]
> variable_continue_on_fail: [true]
> variable_called_party_callgroup: [1]
> variable_dialed_user: [7001]
> variable_dialed_domain: [192.168.2.190]
> variable_call_uuid: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
> variable_originate_disposition: [USER_NOT_REGISTERED]
> variable_max_forwards: [70]
> variable_call_timeout: [20]
> variable_current_application: [info]
>
>
>
> Regards
> Sam
>
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From steveayre at gmail.com  Thu Jun  2 20:19:06 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 2 Jun 2011 17:19:06 +0100
Subject: [Freeswitch-users] how to use mod_mp4
In-Reply-To: <1307025542904-6431457.post@n2.nabble.com>
References: <1306486049569-6410226.post@n2.nabble.com>
	<BANLkTi=4ky6Kj9gE94k_i7tDvdW+LcpjvQ@mail.gmail.com>
	<1306841276439-6422324.post@n2.nabble.com>
	<BANLkTi=O5A_CfCO6PVeyz+RGOCA6Zv1nug@mail.gmail.com>
	<1306858803865-6423457.post@n2.nabble.com>
	<BANLkTinUkgF1semv1VF1edY0WGBsKhJ67Q@mail.gmail.com>
	<1306927048156-6426476.post@n2.nabble.com>
	<BANLkTimLwBeTnNi13zNhZnH9KU4uLWTPwA@mail.gmail.com>
	<1307025542904-6431457.post@n2.nabble.com>
Message-ID: <BANLkTi=BOq1FT-E2FYMSSetDY1S4hTqcFg@mail.gmail.com>

Uh, where did you get that 2nd link from?

That looks like it might be an older version of the library, libmp4 rather
than libmp4v2.

libmp4v2 is the dependancy for mod_mp4, not libmp4

-Steve


On 2 June 2011 15:39, mazilo <Nabble at slickdeals.endjunk.com> wrote:

>
> Steven Ayre wrote:
> >
> > Should do, it's the same library. :)
> I would think so. However, upon a closer look on both
> http://mp4v2.googlecode.com/files/mp4v2-1.9.1.tar.bz2 MP4v2  and
> http://libmp4.svn.sourceforge.net/viewvc/libmp4/trunk/libmp4/ libmp4
> source, it appears MP4v2 uses some different header file names to produce
> libmp4v2.so library while libmp4 produces libmp4_static.a library. Perhaps,
> those who compiled mod_mp4 on a Linux machine already has libmp4 patched to
> produce libmp4v2.so library.
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3
> Watts of electricity.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/how-to-use-mod-mp4-tp6410226p6431457.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From boris at tagnet.ru  Thu Jun  2 20:41:17 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 02 Jun 2011 22:41:17 +0600
Subject: [Freeswitch-users] manipulate CID
In-Reply-To: <BANLkTimWyQ4mzKAWiumtUjwdA4mcw66zUQ@mail.gmail.com>
References: <BANLkTi=ONknfmsocwr+4sQTK54RYUSCTcQ@mail.gmail.com>
	<BANLkTimWyQ4mzKAWiumtUjwdA4mcw66zUQ@mail.gmail.com>
Message-ID: <4DE7BD2D.5070302@tagnet.ru>

Hello!

     Yes, and You're right to set effective_caller_id_number. But have 
You set outbound_caller_id_number before? I see not. And if You need 
just prepend original caller id may be it is better to use
<action appication="set" 
data="effective_caller_id_number=66${caller_id_number}"/> ?
> Is it possible to add prefix to caller id on FS ?
>
> Regds
> Sam
>
>
>
> On Thu, Jun 2, 2011 at 12:18 PM, Sam <u2nsam at gmail.com 
> <mailto:u2nsam at gmail.com>> wrote:
>
>     Hello,
>
>
>     How do i manipulate callerid number ... i have the below statement
>
>     <action application="set"
>     data="effective_caller_id_number=66${outbound_caller_id_number}"/>
>
>     here i want to prefix caller id with 66 so the effective caller id
>     becomes 66 + original callerid .
>
>     using above statement the caller id passed was just 66 to the far end.
>
>     below is the dialplan i used.
>
>     <extension name="7001">
>     <condition field="destination_number" expression="^(7001)"
>     continue="false">
>     <action application="set" data="call_timeout=20"/>
>     <action application="info" />
>     <action application="set"
>     data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED"/>
>
>     <action application="set"
>     data="effective_caller_id_number=66${outbound_caller_id_number}"/>
>     <action application="ring_ready" />
>     <action application="bridge"
>     data="{sip_cid_type=rpid}sofia/gateway/outbound/1646xxxxxxx"/>
>
>     </extension>
>
>
>
>     ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
>
>     This is from header i get .
>
>     From: "" <sip:66 at 203.153.53.138 <mailto:sip%3A66 at 203.153.53.138>>
>
>     -------------------------------------------------------------------------------------------------------------------
>
>
>
>     Dialplan: FreeTDM/1:29/7001 Action info()
>     Dialplan: FreeTDM/1:29/7001 Action
>     set(continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED)
>     Dialplan: FreeTDM/1:29/7001 Action
>     set(effective_caller_id_number=66${outbound_caller_id_number})
>     Dialplan: FreeTDM/1:29/7001 Action ring_ready()
>     Dialplan: FreeTDM/1:29/7001 Action
>     bridge({sip_cid_type=rpid}sofia/gateway/outbound/1646xxxxxxx)
>     2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:119
>     (FreeTDM/1:29/7001) State Change CS_ROUTING -> CS_EXECUTE
>     2011-06-02 06:43:48.111347 [DEBUG] switch_core_session.c:1116 Send
>     signal FreeTDM/1:29/7001 [BREAK]
>     2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:359
>     (FreeTDM/1:29/7001) State ROUTING going to sleep
>     2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:320
>     (FreeTDM/1:29/7001) Running State Change CS_EXECUTE
>     2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:366
>     (FreeTDM/1:29/7001) State EXECUTE
>     2011-06-02 06:43:48.111347 [DEBUG] mod_freetdm.c:451
>     FreeTDM/1:29/7001 CHANNEL EXECUTE
>     2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:157
>     FreeTDM/1:29/7001 Standard EXECUTE
>     EXECUTE FreeTDM/1:29/7001 set(call_timeout=20)
>     2011-06-02 06:43:48.111347 [DEBUG] mod_dptools.c:1059
>     FreeTDM/1:29/7001 SET [call_timeout]=[20]
>     EXECUTE FreeTDM/1:29/7001 info()
>     2011-06-02 06:43:48.111347 [INFO] mod_dptools.c:1202 CHANNEL_DATA:
>     Channel-State: [CS_EXECUTE]
>     Channel-Call-State: [RINGING]
>     Channel-State-Number: [4]
>     Channel-Name: [FreeTDM/1:29/7001]
>     Unique-ID: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
>     Call-Direction: [inbound]
>     Presence-Call-Direction: [inbound]
>     Channel-Call-UUID: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
>     Answer-State: [ringing]
>     Channel-Read-Codec-Name: [PCMA]
>     Channel-Read-Codec-Rate: [8000]
>     Channel-Read-Codec-Bit-Rate: [64000]
>     Channel-Write-Codec-Name: [PCMA]
>     Channel-Write-Codec-Rate: [8000]
>     Channel-Write-Codec-Bit-Rate: [64000]
>     Caller-Direction: [inbound]
>     Caller-Username: [FreeTDM]
>     Caller-Dialplan: [XML]
>     Caller-Caller-ID-Number: [8097328707]
>     Caller-ANI: [8097328707]
>     Caller-Destination-Number: [7001]
>     Caller-Unique-ID: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
>     Caller-Source: [mod_freetdm]
>     Caller-RDNIS: [7001]
>     Caller-Channel-Name: [FreeTDM/1:29/7001]
>     Caller-Profile-Index: [2]
>     Caller-Profile-Created-Time: [1306997028110372]
>     Caller-Channel-Created-Time: [1306997028103773]
>     Caller-Channel-Answered-Time: [0]
>     Caller-Channel-Progress-Time: [0]
>     Caller-Channel-Progress-Media-Time: [0]
>     Caller-Channel-Hangup-Time: [0]
>     Caller-Channel-Transfer-Time: [0]
>     Caller-Screen-Bit: [true]
>     Caller-Privacy-Hide-Name: [false]
>     Caller-Privacy-Hide-Number: [false]
>     variable_direction: [inbound]
>     variable_uuid: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
>     variable_read_codec: [PCMA]
>     variable_read_rate: [8000]
>     variable_write_codec: [PCMA]
>     variable_write_rate: [8000]
>     variable_channel_name: [FreeTDM/1:29/7001]
>     variable_freetdm_span_name: [wp1]
>     variable_freetdm_span_number: [1]
>     variable_freetdm_chan_number: [29]
>     variable_freetdm_bearer_capability: [0]
>     variable_freetdm_bearer_layer1: [3]
>     variable_freetdm_screening_ind: [network-provided]
>     variable_freetdm_presentation_ind: [presentation-allowed]
>     variable_dialed_extension: [7001]
>     variable_transfer_ringback: [local_stream://moh]
>     variable_#: [true]
>     variable_export_vars: [#,*,#]
>     variable_bind_meta_key: [#]
>     variable_hangup_after_bridge: [true]
>     variable_continue_on_fail: [true]
>     variable_called_party_callgroup: [1]
>     variable_dialed_user: [7001]
>     variable_dialed_domain: [192.168.2.190]
>     variable_call_uuid: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
>     variable_originate_disposition: [USER_NOT_REGISTERED]
>     variable_max_forwards: [70]
>     variable_call_timeout: [20]
>     variable_current_application: [info]
>
>
>
>     Regards
>     Sam
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Regards,
Boris


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From anthony.minessale at gmail.com  Thu Jun  2 21:13:47 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 2 Jun 2011 12:13:47 -0500
Subject: [Freeswitch-users] intermittent EXCHANGE_ROUTING_ERROR
In-Reply-To: <1QSAQM-0003GJ-6U@mail.aastral.net>
References: <1QRrkF-0007cj-IK@mail.aastral.net>
	<BANLkTikc5bMokDY6__miju=rOUmsLc7kNg@mail.gmail.com>
	<1QRt7M-0002Zm-GE@mail.aastral.net>
	<BANLkTikXJrxuwUXAd2kYFCiDQOLGQgw9dg@mail.gmail.com>
	<1QS7yM-0005cW-8h@mail.aastral.net>
	<BANLkTi=hv5KZhMUfj1TKbucARH8gzADqpw@mail.gmail.com>
	<1QSAQM-0003GJ-6U@mail.aastral.net>
Message-ID: <BANLkTink0R3qq96hQez786NJhFM_yAzrhg@mail.gmail.com>

then look for a low max-forwards header in the sip traffic

On Thu, Jun 2, 2011 at 11:07 AM, Bill W. <freeswitch at aastral.net> wrote:
>
> Hey Anthony,
>
> All the bridge commands are of the form:
> sofia/profile/18005551212 at 1.2.3.4
>
> Also, if it was looping, would I see that in the logs? ?It would parse
> the dialplan 73 times, correct? ?Because I don't see that.
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From rmorin at blie-ent.com  Thu Jun  2 21:38:54 2011
From: rmorin at blie-ent.com (Rob Morin)
Date: Thu, 2 Jun 2011 13:38:54 -0400
Subject: [Freeswitch-users] Faxing problems if B leg offers T.38
In-Reply-To: <067601cc1afc$b210ada0$163208e0$@blie-ent.com>
References: <1caa01cc197a$5c9ab8d0$15d02a70$@blie-ent.com>	<FB644DB7-59CE-4042-A68B-9811B11FE9B5@ipeva.fr>	<02e001cc19ad$d3d26780$7b773680$@blie-ent.com>	<59ACD9D4-19EE-40D2-8D03-3F798EF08A51@ipeva.fr>	<04b201cc1a21$e8825400$b986fc00$@blie-ent.com>	<060401cc1acd$82adae20$88090a60$@blie-ent.com>	<E92FE87C-23E1-4DEE-9318-3A7D1035B9EE@ipeva.fr>
	<067601cc1afc$b210ada0$163208e0$@blie-ent.com>
Message-ID: <013c01cc214b$f25bad60$d7130820$@blie-ent.com>

Should I open a JIRA case? Two?

Thank you,

Rob

 

From: Rob Morin [mailto:rmorin at blie-ent.com] 
Sent: Wednesday, May 25, 2011 12:56 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] Faxing problems if B leg offers T.38

 

Thank you.

 

So, I believe that there are two potential problems, fixing either of which
will allow me to send faxes.

 

1.       When T.38 is enabled in passthrough mode and the B leg offers T.38,
Freeswitch needs to forward the invite and the responses. Currently it isn?t
forwarding the response back to the B leg.

2.       When T.38 is not enabled and the B leg offers T.38, Freeswitch
needs to respond with a 488 indicating T.38 is not acceptable.  Currently it
doesn?t respond at all.

 

How would I go about making or requesting these fixes?

Thank you,
Rob

 

From: David Ponzone [mailto:david.ponzone at ipeva.fr] 
Sent: Wednesday, May 25, 2011 11:11 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Faxing problems if B leg offers T.38

 

Rob,

 

On my test box, I run a GIT from 05-02, and I don't have any issue.

But I just did the test, and the call flow is different from yours.

 

When I send a fax with my endpoint (Zoiper Softphone), I got the remote fax
tone, and then my Zoiper sends the T38-REINVITE, not the remote gateway.

So I guess there are 2 possible call flows, because I remember in some
circumstances, the remote sends the RE-INVITE.

 

David Ponzone  Direction Technique

email: david.ponzone at ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

 

Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>   -   www.ipeva-studio.com
<BLOCKED::http://www.ipeva-studio.com/> 

 

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

 

 

 

Le 25/05/2011 ? 13:18, Rob Morin a ?crit :

 

Is the fact that I was previously running the snapshot from 25 March
adequate to meet this test?

 

From: Rob Morin [mailto:rmorin at blie-ent.com] 
Sent: Tuesday, May 24, 2011 10:50 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] Faxing problems if B leg offers T.38

 

I can, if someone can tell me where to get the snapshot.  I looked at
files.freeswitch.org and didn?t see one.

 

Prior to updating to the 5-17 snapshot, I was having the same problems with
the 3-25 snapshot that I was using.

 

Thank you,

Rob

 

From: David Ponzone [mailto:david.ponzone at ipeva.fr] 
Sent: Tuesday, May 24, 2011 1:47 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Faxing problems if B leg offers T.38

 

Rob,

 

following Yehavi's comment, could you downgrade your GIT version to a
previous version, like one month old ?

 

David Ponzone  Direction Technique

email: david.ponzone at ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

 

Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>   -   www.ipeva-studio.com
<BLOCKED::http://www.ipeva-studio.com/> 

 

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

 

 

 

Le 24/05/2011 ? 02:59, Rob Morin a ?crit :

 

David,

 

It isn?t the ATA. I?ve tried both the Grandstream HandyTone 502 and
Cisco/Linksys SPA 2102. Same story in both cases.

 

I had my carrier help me troubleshoot and we can see the SIP request from
the carrier come in. When T.38 is enabled, it is forwarded to the A leg, but
the response from the A leg isn?t forwarded back to the B leg. I can produce
traces that show this.

 

When T.38 isn?t enabled, the B leg is sending a T.38 reinvite. FS doesn?t
respond or forward it. So the connection dies.  I have every reason to
believe that I?d be able to send the fax if FS would respond with a 488
indicating T.38 is not acceptable. But instead, all I get is silence. Again,
I can produce traces that demonstrate this.

 

There?s another similar email  trail out there,
http://freeswitch-users.2379917.n2.nabble.com/T-38-via-UPDATE-request-td3821
994.html, but I?m not certain that they?re related.

 

Thank you for your help!
Rob

 

From: David Ponzone [mailto:david.ponzone at ipeva.fr] 
Sent: Monday, May 23, 2011 6:11 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Faxing problems if B leg offers T.38

 

Rob,

 

perhaps you should not consider T38 is 100% interoperable.

You may tell us what ATA is that, because some of them are nice piece of
junk.

 

David Ponzone  Direction Technique

email: david.ponzone at ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

 

Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>   -   www.ipeva-studio.com
<BLOCKED::http://www.ipeva-studio.com/> 

 

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

 





 

Le 23/05/2011 ? 20:51, Rob Morin a ?crit :





I?m having problems faxing if the B leg offers T.38.  Several scenarios, but
always the same result.

 

First, the architecture

 

FAX ? ATA (T.38 optional) ? FS ? Carrier  ? Destination (Possibly another
carrier (PSTN) and the fax machine)

 

On my ATA, T.38 is enabled (Auto Detect).

 

If the destination supports T.38, the Carrier will offer it to Freeswitch.
If the Destination doesn?t support T.38, the B leg doesn?t offer it and the
faxes go through.

 

I?ve set Freeswitch as:

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

      <action application="export" data="t38_passthru=true"/> 

When that?s the case, and the B leg offers T.38, Freeswitch passes through
the offer and the ATA responds, 3 times. But the response never gets sent to
the B leg, so it terminates the call after about 30 seconds.

 

When I set Freeswitch as:

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

The B leg still offers T.38 if its other side is capable. Freeswitch doesn?t
pass the offer through and the B leg terminates the call after about 30
seconds.

 

I?m running on a recent git ( FreeSWITCH Version 1.0.head (git-86d757d
2011-05-17 22-51-47 -0500) ).

 

I?ve had no success attempting to configure FS as a gateway. Frankly, T.38
?passthrough? has had the most success, mostly when the other ends aren?t
T.38 capable and it stays in G711u.

 

Is there something else I need to do to enable T.38 in passthrough mode? 

 

Thank you,
Rob Morin

 

PS ? I can provide tcpdumps of this, or whatever else is necessary.

 

 

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

 

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

 

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

 

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From anthony.minessale at gmail.com  Thu Jun  2 21:57:33 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 2 Jun 2011 12:57:33 -0500
Subject: [Freeswitch-users] Faxing problems if B leg offers T.38
In-Reply-To: <067601cc1afc$b210ada0$163208e0$@blie-ent.com>
References: <1caa01cc197a$5c9ab8d0$15d02a70$@blie-ent.com>
	<FB644DB7-59CE-4042-A68B-9811B11FE9B5@ipeva.fr>
	<02e001cc19ad$d3d26780$7b773680$@blie-ent.com>
	<59ACD9D4-19EE-40D2-8D03-3F798EF08A51@ipeva.fr>
	<04b201cc1a21$e8825400$b986fc00$@blie-ent.com>
	<060401cc1acd$82adae20$88090a60$@blie-ent.com>
	<E92FE87C-23E1-4DEE-9318-3A7D1035B9EE@ipeva.fr>
	<067601cc1afc$b210ada0$163208e0$@blie-ent.com>
Message-ID: <BANLkTik6D=X7OcwySX5g2Xyoc99tqiazVQ@mail.gmail.com>

start by proving them and post them in a jira with full console traces
with sofia global siptrace on and pcaps etc

then wait for us to get to them or email consulting at freeswitch.org for
expedition

On Wed, May 25, 2011 at 11:56 AM, Rob Morin <rmorin at blie-ent.com> wrote:
> Thank you.
>
>
>
> So, I believe that there are two potential problems, fixing either of which
> will allow me to send faxes.
>
>
>
> 1.?????? When T.38 is enabled in passthrough mode and the B leg offers T.38,
> Freeswitch needs to forward the invite and the responses. Currently it isn?t
> forwarding the response back to the B leg.
>
> 2.?????? When T.38 is not enabled and the B leg offers T.38, Freeswitch
> needs to respond with a 488 indicating T.38 is not acceptable.? Currently it
> doesn?t respond at all.
>
>
>
> How would I go about making or requesting these fixes?
>
> Thank you,
> Rob
>
>
>
> From: David Ponzone [mailto:david.ponzone at ipeva.fr]
> Sent: Wednesday, May 25, 2011 11:11 AM
>
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Faxing problems if B leg offers T.38
>
>
>
> Rob,
>
>
>
> On my test box, I run a GIT from 05-02, and I don't have any issue.
>
> But I just did the test, and the call flow is different from yours.
>
>
>
> When I send a fax with my endpoint (Zoiper Softphone), I got the remote fax
> tone, and then my Zoiper sends the T38-REINVITE, not the remote gateway.
>
> So I guess there are 2 possible call flows, because I remember in some
> circumstances, the remote sends the RE-INVITE.
>
>
>
> David Ponzone ?Direction Technique
>
> email: david.ponzone at ipeva.fr
>
> tel: ? ? ?01 74 03 18 97
>
> gsm: ? 06 66 98 76 34
>
>
>
> Service Client?IPeva
>
> tel: ? ? ?0811 46 26 26
>
> www.ipeva.fr? -? ?www.ipeva-studio.com
>
>
>
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
>
> Le 25/05/2011 ? 13:18, Rob Morin a ?crit :
>
> Is the fact that I was previously running the snapshot from 25 March
> adequate to meet this test?
>
>
>
> From:?Rob Morin [mailto:rmorin at blie-ent.com]
> Sent:?Tuesday, May 24, 2011 10:50 AM
> To:?'FreeSWITCH Users Help'
> Subject:?Re: [Freeswitch-users] Faxing problems if B leg offers T.38
>
>
>
> I can, if someone can tell me where to get the snapshot.? I looked
> at?files.freeswitch.org?and didn?t see one.
>
>
>
> Prior to updating to the 5-17 snapshot, I was having the same problems with
> the 3-25 snapshot that I was using.
>
>
>
> Thank you,
>
> Rob
>
>
>
> From:?David Ponzone [mailto:david.ponzone at ipeva.fr]
> Sent:?Tuesday, May 24, 2011 1:47 AM
> To:?FreeSWITCH Users Help
> Subject:?Re: [Freeswitch-users] Faxing problems if B leg offers T.38
>
>
>
> Rob,
>
>
>
> following Yehavi's comment, could you downgrade your GIT version to a
> previous version, like one month old ?
>
>
>
> David Ponzone ?Direction Technique
>
> email:?david.ponzone at ipeva.fr
>
> tel: ? ? ?01 74 03 18 97
>
> gsm: ? 06 66 98 76 34
>
>
>
> Service Client?IPeva
>
> tel: ? ? ?0811 46 26 26
>
> www.ipeva.fr? -? ?www.ipeva-studio.com
>
>
>
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
>
>
>
> Le 24/05/2011 ? 02:59, Rob Morin a ?crit :
>
>
>
> David,
>
>
>
> It isn?t the ATA. I?ve tried both the Grandstream HandyTone 502 and
> Cisco/Linksys SPA 2102. Same story in both cases.
>
>
>
> I had my carrier help me troubleshoot and we can see the SIP request from
> the carrier come in. When T.38 is enabled, it is forwarded to the A leg, but
> the response from the A leg isn?t forwarded back to the B leg. I can produce
> traces that show this.
>
>
>
> When T.38 isn?t enabled, the B leg is sending a T.38 reinvite. FS doesn?t
> respond or forward it. So the connection dies.? I have every reason to
> believe that I?d be able to send the fax if FS would respond with a 488
> indicating T.38 is not acceptable. But instead, all I get is silence. Again,
> I can produce traces that demonstrate this.
>
>
>
> There?s another similar email? trail out
> there,?http://freeswitch-users.2379917.n2.nabble.com/T-38-via-UPDATE-request-td3821994.html,
> but I?m not certain that they?re related.
>
>
>
> Thank you for your help!
> Rob
>
>
>
> From:?David Ponzone [mailto:david.ponzone at ipeva.fr]
> Sent:?Monday, May 23, 2011 6:11 PM
> To:?FreeSWITCH Users Help
> Subject:?Re: [Freeswitch-users] Faxing problems if B leg offers T.38
>
>
>
> Rob,
>
>
>
> perhaps you should not consider T38 is 100% interoperable.
>
> You may tell us what ATA is that, because some of them are nice piece of
> junk.
>
>
>
> David Ponzone ?Direction Technique
>
> email:?david.ponzone at ipeva.fr
>
> tel: ? ? ?01 74 03 18 97
>
> gsm: ? 06 66 98 76 34
>
>
>
> Service Client?IPeva
>
> tel: ? ? ?0811 46 26 26
>
> www.ipeva.fr? -? ?www.ipeva-studio.com
>
>
>
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
>
>
> Le 23/05/2011 ? 20:51, Rob Morin a ?crit :
>
>
> I?m having problems faxing if the B leg offers T.38.? Several scenarios, but
> always the same result.
>
>
>
> First, the architecture
>
>
>
> FAX???ATA (T.38 optional)???FS???Carrier????Destination (Possibly another
> carrier (PSTN) and the fax machine)
>
>
>
> On my ATA, T.38 is enabled (Auto Detect).
>
>
>
> If the destination supports T.38, the Carrier will offer it to Freeswitch.
> If the Destination doesn?t support T.38, the B leg doesn?t offer it and the
> faxes go through.
>
>
>
> I?ve set Freeswitch as:
>
> ??????<action application="set" data="fax_enable_t38=true"/>
>
> ????? <action application="export" data="t38_passthru=true"/>
>
> When that?s the case, and the B leg offers T.38, Freeswitch passes through
> the offer and the ATA responds, 3 times. But the response never gets sent to
> the B leg, so it terminates the call after about 30 seconds.
>
>
>
> When I set Freeswitch as:
>
> ??????<action application="set" data="fax_enable_t38=false"/>
>
> The B leg still offers T.38 if its other side is capable. Freeswitch doesn?t
> pass the offer through and the B leg terminates the call after about 30
> seconds.
>
>
>
> I?m running on a recent git ( FreeSWITCH Version 1.0.head (git-86d757d
> 2011-05-17 22-51-47 -0500) ).
>
>
>
> I?ve had no success attempting to configure FS as a gateway. Frankly, T.38
> ?passthrough? has had the most success, mostly when the other ends aren?t
> T.38 capable and it stays in G711u.
>
>
>
> Is there something else I need to do to enable T.38 in passthrough mode?
>
>
>
> Thank you,
> Rob Morin
>
>
>
> PS ? I can provide tcpdumps of this, or whatever else is necessary.
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
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pstn:+19193869900


From Nabble at slickdeals.endjunk.com  Thu Jun  2 22:40:20 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 2 Jun 2011 11:40:20 -0700 (PDT)
Subject: [Freeswitch-users] how to use mod_mp4
In-Reply-To: <BANLkTi=BOq1FT-E2FYMSSetDY1S4hTqcFg@mail.gmail.com>
References: <1306486049569-6410226.post@n2.nabble.com>
	<BANLkTi=4ky6Kj9gE94k_i7tDvdW+LcpjvQ@mail.gmail.com>
	<1306841276439-6422324.post@n2.nabble.com>
	<BANLkTi=O5A_CfCO6PVeyz+RGOCA6Zv1nug@mail.gmail.com>
	<1306858803865-6423457.post@n2.nabble.com>
	<BANLkTinUkgF1semv1VF1edY0WGBsKhJ67Q@mail.gmail.com>
	<1306927048156-6426476.post@n2.nabble.com>
	<BANLkTimLwBeTnNi13zNhZnH9KU4uLWTPwA@mail.gmail.com>
	<1307025542904-6431457.post@n2.nabble.com>
	<BANLkTi=BOq1FT-E2FYMSSetDY1S4hTqcFg@mail.gmail.com>
Message-ID: <1307040020381-6432408.post@n2.nabble.com>


Steven Ayre wrote:
> 
> Uh, where did you get that 2nd link from?
>From a Google search for libmp4.


That looks like it might be an older version of the library, libmp4 rather
> than libmp4v2.
Indeed. Thanks for clarifying this.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/how-to-use-mod-mp4-tp6410226p6432408.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From leonardo.bidinoto at voicetechnology.com.br  Thu Jun  2 23:33:30 2011
From: leonardo.bidinoto at voicetechnology.com.br (Leonardo P. Bidinoto)
Date: Thu, 2 Jun 2011 16:33:30 -0300
Subject: [Freeswitch-users] Hanged up callers doubt
In-Reply-To: <BANLkTim=ueSmFA_ug2MbbLFMBh0362_2Sg@mail.gmail.com>
References: <BANLkTi=qf4T3Wb9_2TmJdt4Wwif5qipsmg@mail.gmail.com>
	<BANLkTik+DYhPT1ss8-OqpQsvUAA30A5L=Q@mail.gmail.com>
	<BANLkTinxTu=c_9usFNv0sOOWdQ3-DnAYXQ@mail.gmail.com>
	<BANLkTi=g3J1nZTKArouGSJMDHVSE1L0idA@mail.gmail.com>
	<BANLkTi=WN97QMJcpwyEGQNvRxBW1c=UsuQ@mail.gmail.com>
	<BANLkTiky4M2ZgTMS228-DJ6oaEsmkpG19g@mail.gmail.com>
	<BANLkTimzsNxVyFpLf6x8X4E7P58FYwdHng@mail.gmail.com>
	<BANLkTinYynZsX6=zg-yyeP7hEQ7-q5J9+Q@mail.gmail.com>
	<BANLkTim=ueSmFA_ug2MbbLFMBh0362_2Sg@mail.gmail.com>
Message-ID: <BANLkTikV_-=Zkv8zOQVcWyg5Gc06UtRxjw@mail.gmail.com>

Hi,

About this issue, could be a way to destroy the conference from FS, in a
attempt to remove the stucked channels from the "show channels " and
"conference list"?

2011/5/17 Leonardo P. Bidinoto <leonardo.bidinoto at voicetechnology.com.br>

> Sure. Im sending a pcap file made by tcpdump and one that i made by ngrep.
> In both files, it was registering whats happening when i stuck the channel
> by hanging up while using a ESL connection inside a conference(app socket
> 8085 sync full). I did a "conference kick" command in this channel while its
> was waiting to close the ESL connection.
>
>
>
> 2011/5/16 Michael Collins <msc at freeswitch.org>
>
>> Can you tcpdump or otherwise capture the traffic on port 8085? I am
>> curious what is happening with that.
>> -MC
>>
>>
>> On Mon, May 16, 2011 at 12:12 PM, Leonardo P. Bidinoto <
>> leonardo.bidinoto at voicetechnology.com.br> wrote:
>>
>>> hehe, ok michael.
>>>
>>> here is the pastebin link:
>>> http://pastebin.freeswitch.org/16303
>>>
>>>
>>> 2011/5/13 Michael Collins <msc at freeswitch.org>
>>>
>>>> Pastebin this info and select "FreeSWITCH Log" as the syntax
>>>> highlighting. I need the colorized output to read logs. (I'm getting older
>>>> and it's hard for me to ready black and white in an email.)
>>>>
>>>> -MC
>>>>
>>>>
>>>> On Fri, May 13, 2011 at 7:32 AM, Leonardo P. Bidinoto <
>>>> leonardo.bidinoto at voicetechnology.com.br> wrote:
>>>>
>>>>> Hi Michael,
>>>>>
>>>>> Just succeeded to reproduce the problem.
>>>>>
>>>>> The condition is: when a channel inside a conference is using a ESL
>>>>> connection(lets call it "A") through socket application and another ESL
>>>>> connection(lets call it "B") executes a command with this channel, the "B"
>>>>> ESL connection will wait the "A" ESL connection close to execute its
>>>>> command.
>>>>> If the channel hangs up before the "A" ESL connection is closed, then
>>>>> "B" ESL command will never be executed and the stucked channel will still be
>>>>> there, into sofia and the conference too.
>>>>> To verify that, just do "show channels" and "conference list". with
>>>>> "uuid_exists" command, return "false".
>>>>>
>>>>> Here are the actions done by the channel before get stucked:
>>>>>
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.187321
>>>>> [NOTICE] switch_channel.c:816 New Channel sofia/external/
>>>>> 1000123402 at 192.168.0.154 [16e09413-9cb0-4011-a635-f91933a35c0f]
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>>> sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154 entering
>>>>> state [received][100]
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>>> sofia.c:4772 Remote SDP:
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>>> sofia_glue.c:4656 Audio Codec Compare
>>>>> [BV32:107:16000:20:0]/[PCMU:0:8000:20:64000]
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>>> sofia_glue.c:4656 Audio Codec Compare
>>>>> [BV32:107:16000:20:0]/[PCMA:8:8000:20:64000]
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>>> sofia_glue.c:4656 Audio Codec Compare
>>>>> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>>> sofia_glue.c:2788 Set Codec sofia/external/1000123402 at 192.168.0.154PCMU/8000 20 ms 160 samples 64000 bits
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>>> sofia_glue.c:4770 Set 2833 dtmf send/recv payload to 101
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>>> sofia.c:4943 (sofia/external/1000123402 at 192.168.0.154) State Change
>>>>> CS_NEW -> CS_INIT
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>>> switch_core_session.c:1114 Send signal sofia/external/
>>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>>> switch_core_state_machine.c:325 (sofia/external/
>>>>> 1000123402 at 192.168.0.154) Running State Change CS_INIT
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>>> switch_core_state_machine.c:361 (sofia/external/
>>>>> 1000123402 at 192.168.0.154) State INIT
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>>> mod_sofia.c:84 sofia/external/1000123402 at 192.168.0.154 SOFIA INIT
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>>> mod_sofia.c:124 (sofia/external/1000123402 at 192.168.0.154) State Change
>>>>> CS_INIT -> CS_ROUTING
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>>> switch_core_session.c:1114 Send signal sofia/external/
>>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>>> switch_core_state_machine.c:361 (sofia/external/
>>>>> 1000123402 at 192.168.0.154) State INIT going to sleep
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>>> switch_core_state_machine.c:325 (sofia/external/
>>>>> 1000123402 at 192.168.0.154) Running State Change CS_ROUTING
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>>> switch_channel.c:1665 (sofia/external/1000123402 at 192.168.0.154)
>>>>> Callstate Change DOWN -> RINGING
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>>> switch_core_state_machine.c:364 (sofia/external/
>>>>> 1000123402 at 192.168.0.154) State ROUTING
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>>> mod_sofia.c:147 sofia/external/1000123402 at 192.168.0.154 SOFIA ROUTING
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG]
>>>>> switch_core_state_machine.c:77 sofia/external/1000123402 at 192.168.0.154Standard ROUTING
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [INFO]
>>>>> mod_dialplan_xml.c:331 Processing Phone1 <1000123402>->1234567890 in context
>>>>> public
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>> 1000123402 at 192.168.0.154 parsing [public->unloop] continue=false
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>> 1000123402 at 192.168.0.154 Regex (PASS) [unloop] ${unroll_loops}(true)
>>>>> =~ /^true$/ break=on-false
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>> 1000123402 at 192.168.0.154 Regex (FAIL) [unloop] ${sip_looped_call}() =~
>>>>> /^true$/ break=on-false
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>> 1000123402 at 192.168.0.154 parsing [public->public_extensions]
>>>>> continue=false
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>> 1000123402 at 192.168.0.154 Regex (PASS) [public_extensions]
>>>>> destination_number(1234567890) =~ /^(\d*)$/ break=on-false
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>> 1000123402 at 192.168.0.154 Action transfer(1234567890 XML default)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
>>>>> switch_core_state_machine.c:119 (sofia/external/
>>>>> 1000123402 at 192.168.0.154) State Change CS_ROUTING -> CS_EXECUTE
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
>>>>> switch_core_session.c:1114 Send signal sofia/external/
>>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
>>>>> switch_core_state_machine.c:364 (sofia/external/
>>>>> 1000123402 at 192.168.0.154) State ROUTING going to sleep
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
>>>>> switch_core_state_machine.c:325 (sofia/external/
>>>>> 1000123402 at 192.168.0.154) Running State Change CS_EXECUTE
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
>>>>> switch_core_state_machine.c:371 (sofia/external/
>>>>> 1000123402 at 192.168.0.154) State EXECUTE
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG]
>>>>> mod_sofia.c:240 sofia/external/1000123402 at 192.168.0.154 SOFIA EXECUTE
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>>>>> switch_core_state_machine.c:157 sofia/external/
>>>>> 1000123402 at 192.168.0.154 Standard EXECUTE
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>> 1000123402 at 192.168.0.154 transfer(1234567890 XML default)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>>>>> switch_ivr.c:1597 (sofia/external/1000123402 at 192.168.0.154) State
>>>>> Change CS_EXECUTE -> CS_ROUTING
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>>>>> switch_core_session.c:1114 Send signal sofia/external/
>>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>>>>> switch_core_session.c:707 Send signal sofia/external/
>>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327
>>>>> [NOTICE] switch_ivr.c:1603 Transfer sofia/external/
>>>>> 1000123402 at 192.168.0.154 to XML[1234567890 at default]
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>>>>> switch_core_state_machine.c:371 (sofia/external/
>>>>> 1000123402 at 192.168.0.154) State EXECUTE going to sleep
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>>>>> switch_core_state_machine.c:325 (sofia/external/
>>>>> 1000123402 at 192.168.0.154) Running State Change CS_ROUTING
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>>>>> switch_core_state_machine.c:364 (sofia/external/
>>>>> 1000123402 at 192.168.0.154) State ROUTING
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>>>>> mod_sofia.c:147 sofia/external/1000123402 at 192.168.0.154 SOFIA ROUTING
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG]
>>>>> switch_core_state_machine.c:77 sofia/external/1000123402 at 192.168.0.154Standard ROUTING
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [INFO]
>>>>> mod_dialplan_xml.c:331 Processing Phone1 <1000123402>->1234567890 in context
>>>>> default
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>> 1000123402 at 192.168.0.154 parsing [default->flex] continue=false
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>> 1000123402 at 192.168.0.154 Regex (PASS) [flex]
>>>>> destination_number(1234567890) =~ /^(\d+)$/ break=on-false
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>> 1000123402 at 192.168.0.154 Action log(INFO VOICE received
>>>>> dest=1234567890)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>> 1000123402 at 192.168.0.154 Action set(playback_terminators=#)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>> 1000123402 at 192.168.0.154 Action log(INFO Let's do some ivrd, shall
>>>>> we?)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>> 1000123402 at 192.168.0.154 Action
>>>>> set(ivr_path=/usr/local/ipconf/fs/daemon/ivr.rb)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>> 1000123402 at 192.168.0.154 Action socket(localhost:8084 full)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
>>>>> switch_core_state_machine.c:119 (sofia/external/
>>>>> 1000123402 at 192.168.0.154) State Change CS_ROUTING -> CS_EXECUTE
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
>>>>> switch_core_session.c:1114 Send signal sofia/external/
>>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
>>>>> switch_core_state_machine.c:364 (sofia/external/
>>>>> 1000123402 at 192.168.0.154) State ROUTING going to sleep
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
>>>>> switch_core_state_machine.c:325 (sofia/external/
>>>>> 1000123402 at 192.168.0.154) Running State Change CS_EXECUTE
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
>>>>> switch_core_state_machine.c:371 (sofia/external/
>>>>> 1000123402 at 192.168.0.154) State EXECUTE
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
>>>>> mod_sofia.c:240 sofia/external/1000123402 at 192.168.0.154 SOFIA EXECUTE
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG]
>>>>> switch_core_state_machine.c:157 sofia/external/
>>>>> 1000123402 at 192.168.0.154 Standard EXECUTE
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>> 1000123402 at 192.168.0.154 log(INFO VOICE received dest=1234567890)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [INFO]
>>>>> mod_dptools.c:1184 VOICE received dest=1234567890
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>> 1000123402 at 192.168.0.154 set(playback_terminators=#)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331 [DEBUG]
>>>>> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
>>>>> [playback_terminators]=[#]
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>> 1000123402 at 192.168.0.154 log(INFO Let's do some ivrd, shall we?)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331 [INFO]
>>>>> mod_dptools.c:1184 Let's do some ivrd, shall we?
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>> 1000123402 at 192.168.0.154set(ivr_path=/usr/local/ipconf/fs/daemon/ivr.rb)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331 [DEBUG]
>>>>> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
>>>>> [ivr_path]=[/usr/local/ipconf/fs/daemon/ivr.rb]
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>> 1000123402 at 192.168.0.154 socket(localhost:8084 full)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309 [DEBUG]
>>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>>> Execute answer()
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>> 1000123402 at 192.168.0.154 answer()
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309 [DEBUG]
>>>>> sofia_glue.c:3022 AUDIO RTP [sofia/external/1000123402 at 192.168.0.154]
>>>>> 192.168.0.154 port 24232 -> 192.168.0.111 port 4046 codec: 0 ms: 20
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309 [DEBUG]
>>>>> switch_rtp.c:1623 Starting timer [soft] 160 bytes per 20ms
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
>>>>> sofia_glue.c:3284 Set 2833 dtmf send payload to 101
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
>>>>> sofia_glue.c:3289 Set 2833 dtmf receive payload to 101
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
>>>>> mod_sofia.c:681 Local SDP sofia/external/1000123402 at 192.168.0.154:
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
>>>>> switch_core_session.c:707 Send signal sofia/external/
>>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG]
>>>>> switch_channel.c:2827 (sofia/external/1000123402 at 192.168.0.154)
>>>>> Callstate Change RINGING -> ACTIVE
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313
>>>>> [NOTICE] mod_dptools.c:930 Channel [sofia/external/
>>>>> 1000123402 at 192.168.0.154] has been answered
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.933312 [DEBUG]
>>>>> sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154 entering
>>>>> state [completed][200]
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.040285 [DEBUG]
>>>>> sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154 entering
>>>>> state [ready][200]
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.082294 [DEBUG]
>>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>>> Execute
>>>>> playback(/usr/local/freeswitch/sounds/flex/app32/teste/WelcomeToTheConferenceSystem.wav)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>> 1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/WelcomeToTheConferenceSystem.wav)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.082294 [DEBUG]
>>>>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.114277 [DEBUG]
>>>>> switch_ivr_play_say.c:1649 done playing file
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.155281 [DEBUG]
>>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>>> Execute read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Mainmenu.wav
>>>>> flex_digits 5000 )
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>> 1000123402 at 192.168.0.154 read(1 1
>>>>> /usr/local/freeswitch/sounds/flex/app32/teste/Mainmenu.wav flex_digits 5000
>>>>> )
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.155281 [DEBUG]
>>>>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.893670 [DEBUG]
>>>>> switch_rtp.c:3280 RTP RECV DTMF 1:1120
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.893670 [DEBUG]
>>>>> switch_ivr_play_say.c:1649 done playing file
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.897674 [DEBUG]
>>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>>> Execute set(flex_digits)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>> 1000123402 at 192.168.0.154 set(flex_digits)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.897674 [DEBUG]
>>>>> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
>>>>> [flex_digits]=[UNDEF]
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.940673 [DEBUG]
>>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>>> Execute read(1 1
>>>>> /usr/local/freeswitch/sounds/flex/app32/teste/PleaseEnterYourPassCodeNow.wav
>>>>> flex_digits 5000 )
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>> 1000123402 at 192.168.0.154 read(1 1
>>>>> /usr/local/freeswitch/sounds/flex/app32/teste/PleaseEnterYourPassCodeNow.wav
>>>>> flex_digits 5000 )
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.940673 [DEBUG]
>>>>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.113662 [DEBUG]
>>>>> switch_rtp.c:3280 RTP RECV DTMF 8:640
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.113662 [DEBUG]
>>>>> switch_ivr_play_say.c:1649 done playing file
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.116663 [DEBUG]
>>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>>> Execute set(flex_digits)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>> 1000123402 at 192.168.0.154 set(flex_digits)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.116663 [DEBUG]
>>>>> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
>>>>> [flex_digits]=[UNDEF]
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.117663 [DEBUG]
>>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>>> Execute read(1 11 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav
>>>>> flex_digits 5000 #,*)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>> 1000123402 at 192.168.0.154 read(1 11
>>>>> /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 5000
>>>>> #,*)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.117663 [DEBUG]
>>>>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.153658 [DEBUG]
>>>>> switch_ivr_play_say.c:1649 done playing file
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.053615 [DEBUG]
>>>>> switch_rtp.c:3280 RTP RECV DTMF #:960
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.055618 [DEBUG]
>>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>>> Execute set(flex_digits)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>> 1000123402 at 192.168.0.154 set(flex_digits)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.055618 [DEBUG]
>>>>> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
>>>>> [flex_digits]=[UNDEF]
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.146613 [DEBUG]
>>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>>> Execute
>>>>> playback(/usr/local/freeswitch/sounds/flex/app32/teste/InternalHelp.wav)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>> 1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/InternalHelp.wav)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.146613 [DEBUG]
>>>>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603 [DEBUG]
>>>>> switch_rtp.c:3280 RTP RECV DTMF #:800
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603 [DEBUG]
>>>>> mod_dptools.c:1664 Digit #
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603 [DEBUG]
>>>>> switch_ivr_play_say.c:1649 done playing file
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.350617 [DEBUG]
>>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>>> Execute
>>>>> playback(/usr/local/freeswitch/sounds/flex/app32/teste/YouAreTheOnlyPersonOnThisConference.wav)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>> 1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/YouAreTheOnlyPersonOnThisConference.wav)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.350617 [DEBUG]
>>>>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.253520 [DEBUG]
>>>>> switch_ivr_play_say.c:1649 done playing file
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.322510 [DEBUG]
>>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>>> Execute conference(15646 at teste+flags{waste})
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>> 1000123402 at 192.168.0.154 conference(15646 at teste+flags{waste})
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492 [DEBUG]
>>>>> mod_conference.c:5582 Raw Codec Activation Success L16 at 8000hz 1
>>>>> channel 20ms
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492 [DEBUG]
>>>>> mod_conference.c:5627 Raw Codec Activation Success L16 at 8000hz 1
>>>>> channel 20ms
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492 [DEBUG]
>>>>> switch_core_codec.c:116 sofia/external/1000123402 at 192.168.0.154 Push
>>>>> codec L16:70
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:05.280469 [DEBUG]
>>>>> switch_core_session.c:707 Send signal sofia/external/
>>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:05.280469 [DEBUG]
>>>>> mod_conference.c:2557 Setup timer soft success interval: 20  samples: 160
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.573928 [DEBUG]
>>>>> switch_rtp.c:3280 RTP RECV DTMF *:960
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.593929
>>>>> [DEBUG]  mod_conference.c:2021 Execute app: socket, localhost:8085 sync full
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.593929 [DEBUG]
>>>>> switch_core_codec.c:141 sofia/external/1000123402 at 192.168.0.154Restore previous codec PCMU:0.
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>> 1000123402 at 192.168.0.154 socket(localhost:8085 sync full)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.097927 [DEBUG]
>>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>>> Execute read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav
>>>>> flex_digits 2000 )
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>> 1000123402 at 192.168.0.154 read(1 1
>>>>> /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 )
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.097927 [DEBUG]
>>>>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.133923 [DEBUG]
>>>>> switch_ivr_play_say.c:1649 done playing file
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.373913 [DEBUG]
>>>>> switch_rtp.c:3280 RTP RECV DTMF 1:1120
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.375912 [DEBUG]
>>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>>> Execute set(flex_digits)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>> 1000123402 at 192.168.0.154 set(flex_digits)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.375912 [DEBUG]
>>>>> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
>>>>> [flex_digits]=[UNDEF]
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.490911 [DEBUG]
>>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>>> Execute
>>>>> playback(/usr/local/freeswitch/sounds/flex/app32/teste/SelfMuteOn.wav)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>> 1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/SelfMuteOn.wav)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.490911 [DEBUG]
>>>>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.233824 [DEBUG]
>>>>> switch_ivr_play_say.c:1649 done playing file
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.235828 [DEBUG]
>>>>> switch_cpp.cpp:988 sofia/external/1000123402 at 192.168.0.154destroy/unlink session from object
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.235828 [DEBUG]
>>>>> switch_core_codec.c:116 sofia/external/1000123402 at 192.168.0.154 Push
>>>>> codec L16:70
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.433668 [DEBUG]
>>>>> switch_rtp.c:3280 RTP RECV DTMF *:800
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.453667
>>>>> [DEBUG]  mod_conference.c:2021 Execute app: socket, localhost:8085 sync full
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.453667 [DEBUG]
>>>>> switch_core_codec.c:141 sofia/external/1000123402 at 192.168.0.154Restore previous codec PCMU:0.
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>> 1000123402 at 192.168.0.154 socket(localhost:8085 sync full)
>>>>>
>>>>> ====================================================================================================================================================
>>>>> While Inside this connection, a "conference 15646 kick [member_id of
>>>>> this channels]" command is executed by a fs_cli console and get stuck while
>>>>> waiting response.
>>>>>
>>>>> ====================================================================================================================================================
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.000850 [DEBUG]
>>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>>> Execute read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav
>>>>> flex_digits 2000 )
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>> 1000123402 at 192.168.0.154 read(1 1
>>>>> /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 )
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.000850 [DEBUG]
>>>>> switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.033846 [DEBUG]
>>>>> switch_ivr_play_say.c:1649 done playing file
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.553821 [DEBUG]
>>>>> switch_rtp.c:3280 RTP RECV DTMF 1:960
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.555825 [DEBUG]
>>>>> switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command
>>>>> Execute set(flex_digits)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>> 1000123402 at 192.168.0.154 set(flex_digits)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.555825 [DEBUG]
>>>>> mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET
>>>>> [flex_digits]=[UNDEF]
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [DEBUG]
>>>>> switch_channel.c:2560 (sofia/external/1000123402 at 192.168.0.154)
>>>>> Callstate Change ACTIVE -> HANGUP
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244
>>>>> [NOTICE] sofia.c:538 Hangup sofia/external/1000123402 at 192.168.0.154[CS_EXECUTE] [NORMAL_CLEARING]
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [DEBUG]
>>>>> switch_channel.c:2576 Send signal sofia/external/
>>>>> 1000123402 at 192.168.0.154 [KILL]
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [DEBUG]
>>>>> switch_core_session.c:1114 Send signal sofia/external/
>>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
>>>>> switch_core_session.c:2057 sofia/external/1000123402 at 192.168.0.154skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
>>>>> switch_cpp.cpp:988 sofia/external/1000123402 at 192.168.0.154destroy/unlink session from object
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
>>>>> switch_core_session.c:2057 sofia/external/1000123402 at 192.168.0.154skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
>>>>> switch_core_codec.c:116 sofia/external/1000123402 at 192.168.0.154 Push
>>>>> codec L16:70
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG]
>>>>> mod_conference.c:2815 Channel leaving conference, cause: NORMAL_CLEARING
>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.757239 [DEBUG]
>>>>> mod_conference.c:6104 sofia/external/1000123402 at 192.168.0.154 skip
>>>>> receive message [UNBRIDGE] (channel is hungup already)
>>>>>
>>>>> I hope this info helps.
>>>>>
>>>>> 2011/5/12 Michael Collins <msc at freeswitch.org>
>>>>>
>>>>>>
>>>>>>
>>>>>> On Thu, May 12, 2011 at 12:27 PM, Leonardo P. Bidinoto <
>>>>>> leonardo.bidinoto at voicetechnology.com.br> wrote:
>>>>>>
>>>>>>> Hi Michael,
>>>>>>>
>>>>>>> Im not using to any cdr module.
>>>>>>
>>>>>>
>>>>>> I would recommend that you do several things:
>>>>>>
>>>>>> #1 - update to latest git
>>>>>> #2 - rotate logs
>>>>>> #3 - enable uuid logging (see logfile.conf.xml - param name "uuid")
>>>>>> #4 - reproduce the symptom with a single call (if possible)
>>>>>> #5 - pastebin the log for the uuid in question and link to it in this
>>>>>> thread
>>>>>>
>>>>>> From there hopefully we'll get a clue as to what is happening.
>>>>>> -MC
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Leonardo Pires Bidinoto
>>>>> Voice Technology
>>>>> www.voicetechnology.com.br
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Leonardo Pires Bidinoto
>>> Voice Technology
>>> www.voicetechnology.com.br
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Leonardo Pires Bidinoto
> Voice Technology
> www.voicetechnology.com.br
>



-- 
Leonardo Pires Bidinoto
Voice Technology
www.voicetechnology.com.br
-------------- next part --------------
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From mgende at gendesign.com  Thu Jun  2 23:49:18 2011
From: mgende at gendesign.com (Michael Gende)
Date: Thu, 2 Jun 2011 14:49:18 -0500
Subject: [Freeswitch-users] ptime mismatch confusion
Message-ID: <BANLkTim-3A4s9G3eLvHTEssg8=xf=-ysCA@mail.gmail.com>

Hello,

If you are a codec/ptime person, great. I've been "round and round" with the
vendor on this and and not making much headway.

We've an FS switch that has two numbers registered with our provider. That
provider is, in turn, getting service from another gateway provider of their
own.

On our FS, one DID works with no issues and has for more than a year. They
other recently started having ptime mismatch issues making it unusable.
Tried Scrooge setting to no avail.

For the daring (and merciful), here's FS output for a "bad" call and then a
"good" one (just a little output from the start):


2011-06-02 14:12:06.756170 [DEBUG] switch_core_state_machine.c:397
(sofia/external/8151231234 at xxx.xxx.xxx.xxx:5060<http://8154835414 at 208.94.159.10:5060/>)
Running State Change CS_NEW
 2011-06-02 14:12:06.756170 [DEBUG] sofia.c:3210 Channel sofia/external/
8151231234 at xxx.xxx.xxx.xxx:5060
<http://8154835414 at 208.94.159.10:5060/>entering state [received][100]
2011-06-02 14:12:06.756170 [DEBUG] sofia.c:3217 Remote SDP:


v=0


o=Acme_UAS 0 1 IN IP4 xxx.xxx.xxx.xxx


s=SIP Media Capabilities


c=IN IP4 yyy.yyy.yyy.yyy


t=0 0


m=audio 52024 RTP/AVP 0 18 101


a=rtpmap:0 PCMU/8000


a=rtpmap:18 G729/8000


a=rtpmap:101 telephone-event/8000


*a=maxptime:30     *





2011-06-02 14:12:06.756170 [DEBUG] switch_core_state_machine.c:403
(sofia/external/8151231234 at xxx.xxx.xxx.xxx:5060<http://8154835414 at 208.94.159.10:5060/>)
State NEW
2011-06-02 14:12:06.756170 [DEBUG] sofia_glue.c:3081 *Audio Codec Compare
[PCMU:0:8000:30]/[PCMU:0:8000:**20]*

2011-06-02 14:12:06.756170 [DEBUG] sofia_glue.c:3092 Bah HUMBUG! Sticking
with PCMU at 8000h@20i

2011-06-02 14:12:06.756170 [DEBUG] sofia_glue.c:2039 Set Codec
sofia/external/8151231234 at xxx.xxx.xxx.xxx:5060<http://8154835414 at 208.94.159.10:5060/>PCMU/8000
20 ms 160 samples
2011-06-02 14:12:06.756170 [DEBUG] sofia_glue.c:3041 Set 2833 dtmf payload
to 101



Note that whilst it says its sticking with ptime of 20 (bolded above),* our
vendor in-the-middle sees us as still using ptime=30*. Weird. But, on the
same switch with different DID, we get the following "good" result:



2011-06-02 14:15:15.135193 [DEBUG] switch_core_state_machine.c:397
(sofia/external/8151231234 at xxx.xxx.xxx.xxx:5060<http://8159535620 at 208.94.157.10:5060/>)
Running State Change CS_NEW
 2011-06-02 14:15:15.135193 [DEBUG] sofia.c:3210 Channel
sofia/external/8151231234@
xxx.xxx.xxx.xxx:5060 <http://8159535620 at 208.94.157.10:5060/> entering state
[received][100]
2011-06-02 14:15:15.135193 [DEBUG] sofia.c:3217 Remote SDP:


v=0


o=Acme_UAS 0 1 IN IP4 xxx.xxx.xxx.xxx


s=SIP Media Capabilities


c=IN IP4 yyy.yyy.yyy.yyy


t=0 0


m=audio 52052 RTP/AVP 0 18 101


a=rtpmap:0 PCMU/8000


a=rtpmap:18 G729/8000


a=rtpmap:101 telephone-event/8000


*a=maxptime:20*





2011-06-02 14:15:15.135193 [DEBUG] switch_core_state_machine.c:403
(sofia/external/8151231234 at xxx.xxx.xxx.xxx.10:5060<http://8159535620 at 208.94.157.10:5060/>)
State NEW
2011-06-02 14:15:15.135193 [DEBUG] sofia_glue.c:3081 *Audio Codec Compare
[PCMU:0:8000:20]/[PCMU:0:8000:**20]*

2011-06-02 14:15:15.135193 [DEBUG] sofia_glue.c:3092 Bah HUMBUG! Sticking
with PCMU at 8000h@20i

2011-06-02 14:15:15.135193 [DEBUG] sofia_glue.c:2039 Set Codec
sofia/external/8151231234 at xxx.xxx.xxx.xxx:5060<http://8159535620 at 208.94.157.10:5060/>PCMU/8000
20 ms 160 samples
2011-06-02 14:15:15.135193 [DEBUG] sofia_glue.c:3041 Set 2833 dtmf payload
to 101


Naturally, the "far end" vendor says they think its a "FreeSwitch problem".
Huh? For my part, I'd just like to FORCE a ptime of 20 in all cases, no
matter the codec, for incoming calls, either DID. Tried, but so far, no go.

A packet capture showed "bad" call coming in with ptime max of 30, we
respond that 30 is OK but them re-invite with ptime set to 20, but it
doesn't seem to "catch". On the good DID, they proffer ptime max of 30, we
respond with ptime of 20 and life is good.

Sorry to post again on this. I have put in some hours but still same results
so far. Any suggestions much appreciated.

Regards,

Mike
-------------- next part --------------
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From anthony.minessale at gmail.com  Thu Jun  2 23:55:14 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 2 Jun 2011 14:55:14 -0500
Subject: [Freeswitch-users] ptime mismatch confusion
In-Reply-To: <BANLkTim-3A4s9G3eLvHTEssg8=xf=-ysCA@mail.gmail.com>
References: <BANLkTim-3A4s9G3eLvHTEssg8=xf=-ysCA@mail.gmail.com>
Message-ID: <BANLkTimd3fyz3S0ATTTSq19gM+Nthdtdzg@mail.gmail.com>

Thats not really enough info
you should have a full pcap of a single call

also you should not be using that codec negotiation scrooge unless its
a very specific case.

get a trace without the scrooge mode and consider reporting it to
consulting at freeswitch.org for expedited commercial assistance.



On Thu, Jun 2, 2011 at 2:49 PM, Michael Gende <mgende at gendesign.com> wrote:
> Hello,
>
> If you are a codec/ptime person, great. I've been "round and round" with the
> vendor on this and and not making much headway.
>
> We've an FS switch that has two numbers registered with our provider. That
> provider is, in turn, getting service from another gateway provider of their
> own.
>
> On our FS, one DID works with no issues and has for more than a year. They
> other recently started having ptime mismatch issues making it unusable.
> Tried Scrooge setting to no avail.
>
> For the daring (and merciful), here's FS output for a "bad" call and then a
> "good" one (just a little output from the start):
>
>
> 2011-06-02 14:12:06.756170 [DEBUG] switch_core_state_machine.c:397
> (sofia/external/8151231234 at xxx.xxx.xxx.xxx:5060) Running State Change CS_NEW
>
> 2011-06-02 14:12:06.756170 [DEBUG] sofia.c:3210 Channel
> sofia/external/8151231234 at xxx.xxx.xxx.xxx:5060 entering state
> [received][100]
> 2011-06-02 14:12:06.756170 [DEBUG] sofia.c:3217 Remote SDP:
>
>
> v=0
>
>
> o=Acme_UAS 0 1 IN IP4 xxx.xxx.xxx.xxx
>
>
> s=SIP Media Capabilities
>
>
> c=IN IP4 yyy.yyy.yyy.yyy
>
>
> t=0 0
>
>
> m=audio 52024 RTP/AVP 0 18 101
>
>
> a=rtpmap:0 PCMU/8000
>
>
> a=rtpmap:18 G729/8000
>
>
> a=rtpmap:101 telephone-event/8000
>
>
> a=maxptime:30
>
>
>
>
>
> 2011-06-02 14:12:06.756170 [DEBUG] switch_core_state_machine.c:403
> (sofia/external/8151231234 at xxx.xxx.xxx.xxx:5060) State NEW
>
> 2011-06-02 14:12:06.756170 [DEBUG] sofia_glue.c:3081 Audio Codec Compare
> [PCMU:0:8000:30]/[PCMU:0:8000:20]
>
> 2011-06-02 14:12:06.756170 [DEBUG] sofia_glue.c:3092 Bah HUMBUG! Sticking
> with PCMU at 8000h@20i
>
> 2011-06-02 14:12:06.756170 [DEBUG] sofia_glue.c:2039 Set Codec
> sofia/external/8151231234 at xxx.xxx.xxx.xxx:5060 PCMU/8000 20 ms 160 samples
>
> 2011-06-02 14:12:06.756170 [DEBUG] sofia_glue.c:3041 Set 2833 dtmf payload
> to 101
>
>
>
> Note that whilst it says its sticking with ptime of 20 (bolded above), our
> vendor in-the-middle sees us as still using ptime=30. Weird. But, on the
> same switch with different DID, we get the following "good" result:
>
>
>
> 2011-06-02 14:15:15.135193 [DEBUG] switch_core_state_machine.c:397
> (sofia/external/8151231234 at xxx.xxx.xxx.xxx:5060) Running State Change CS_NEW
>
> 2011-06-02 14:15:15.135193 [DEBUG] sofia.c:3210 Channel
> sofia/external/8151231234@ xxx.xxx.xxx.xxx:5060 entering state
> [received][100]
> 2011-06-02 14:15:15.135193 [DEBUG] sofia.c:3217 Remote SDP:
>
>
> v=0
>
>
> o=Acme_UAS 0 1 IN IP4 xxx.xxx.xxx.xxx
>
>
> s=SIP Media Capabilities
>
>
> c=IN IP4 yyy.yyy.yyy.yyy
>
>
> t=0 0
>
>
> m=audio 52052 RTP/AVP 0 18 101
>
>
> a=rtpmap:0 PCMU/8000
>
>
> a=rtpmap:18 G729/8000
>
>
> a=rtpmap:101 telephone-event/8000
>
>
> a=maxptime:20
>
>
>
>
>
> 2011-06-02 14:15:15.135193 [DEBUG] switch_core_state_machine.c:403
> (sofia/external/8151231234 at xxx.xxx.xxx.xxx.10:5060) State NEW
>
> 2011-06-02 14:15:15.135193 [DEBUG] sofia_glue.c:3081 Audio Codec Compare
> [PCMU:0:8000:20]/[PCMU:0:8000:20]
>
> 2011-06-02 14:15:15.135193 [DEBUG] sofia_glue.c:3092 Bah HUMBUG! Sticking
> with PCMU at 8000h@20i
>
> 2011-06-02 14:15:15.135193 [DEBUG] sofia_glue.c:2039 Set Codec
> sofia/external/8151231234 at xxx.xxx.xxx.xxx:5060 PCMU/8000 20 ms 160 samples
>
> 2011-06-02 14:15:15.135193 [DEBUG] sofia_glue.c:3041 Set 2833 dtmf payload
> to 101
>
>
> Naturally, the "far end" vendor says they think its a "FreeSwitch problem".
> Huh? For my part, I'd just like to FORCE a ptime of 20 in all cases, no
> matter the codec, for incoming calls, either DID. Tried, but so far, no go.
>
> A packet capture showed "bad" call coming in with ptime max of 30, we
> respond that 30 is OK but them re-invite with ptime set to 20, but it
> doesn't seem to "catch". On the good DID, they proffer ptime max of 30, we
> respond with ptime of 20 and life is good.
>
> Sorry to post again on this. I have put in some hours but still same results
> so far. Any suggestions much appreciated.
>
> Regards,
>
> Mike
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From mgende at gendesign.com  Thu Jun  2 23:57:47 2011
From: mgende at gendesign.com (Michael Gende)
Date: Thu, 2 Jun 2011 14:57:47 -0500
Subject: [Freeswitch-users] ptime mismatch confusion
In-Reply-To: <BANLkTimd3fyz3S0ATTTSq19gM+Nthdtdzg@mail.gmail.com>
References: <BANLkTim-3A4s9G3eLvHTEssg8=xf=-ysCA@mail.gmail.com>
	<BANLkTimd3fyz3S0ATTTSq19gM+Nthdtdzg@mail.gmail.com>
Message-ID: <BANLkTin3HpV-7UnCUiiHLoYHSQBFksx4hQ@mail.gmail.com>

That's what I'll try. Thanks.

On Thu, Jun 2, 2011 at 2:55 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Thats not really enough info
> you should have a full pcap of a single call
>
> also you should not be using that codec negotiation scrooge unless its
> a very specific case.
>
> get a trace without the scrooge mode and consider reporting it to
> consulting at freeswitch.org for expedited commercial assistance.
>
>
>
> On Thu, Jun 2, 2011 at 2:49 PM, Michael Gende <mgende at gendesign.com>
> wrote:
> > Hello,
> >
> > If you are a codec/ptime person, great. I've been "round and round" with
> the
> > vendor on this and and not making much headway.
> >
> > We've an FS switch that has two numbers registered with our provider.
> That
> > provider is, in turn, getting service from another gateway provider of
> their
> > own.
> >
> > On our FS, one DID works with no issues and has for more than a year.
> They
> > other recently started having ptime mismatch issues making it unusable.
> > Tried Scrooge setting to no avail.
> >
> > For the daring (and merciful), here's FS output for a "bad" call and then
> a
> > "good" one (just a little output from the start):
> >
> >
> > 2011-06-02 14:12:06.756170 [DEBUG] switch_core_state_machine.c:397
> > (sofia/external/8151231234 at xxx.xxx.xxx.xxx:5060) Running State Change
> CS_NEW
> >
> > 2011-06-02 14:12:06.756170 [DEBUG] sofia.c:3210 Channel
> > sofia/external/8151231234 at xxx.xxx.xxx.xxx:5060 entering state
> > [received][100]
> > 2011-06-02 14:12:06.756170 [DEBUG] sofia.c:3217 Remote SDP:
> >
> >
> > v=0
> >
> >
> > o=Acme_UAS 0 1 IN IP4 xxx.xxx.xxx.xxx
> >
> >
> > s=SIP Media Capabilities
> >
> >
> > c=IN IP4 yyy.yyy.yyy.yyy
> >
> >
> > t=0 0
> >
> >
> > m=audio 52024 RTP/AVP 0 18 101
> >
> >
> > a=rtpmap:0 PCMU/8000
> >
> >
> > a=rtpmap:18 G729/8000
> >
> >
> > a=rtpmap:101 telephone-event/8000
> >
> >
> > a=maxptime:30
> >
> >
> >
> >
> >
> > 2011-06-02 14:12:06.756170 [DEBUG] switch_core_state_machine.c:403
> > (sofia/external/8151231234 at xxx.xxx.xxx.xxx:5060) State NEW
> >
> > 2011-06-02 14:12:06.756170 [DEBUG] sofia_glue.c:3081 Audio Codec Compare
> > [PCMU:0:8000:30]/[PCMU:0:8000:20]
> >
> > 2011-06-02 14:12:06.756170 [DEBUG] sofia_glue.c:3092 Bah HUMBUG! Sticking
> > with PCMU at 8000h@20i
> >
> > 2011-06-02 14:12:06.756170 [DEBUG] sofia_glue.c:2039 Set Codec
> > sofia/external/8151231234 at xxx.xxx.xxx.xxx:5060 PCMU/8000 20 ms 160
> samples
> >
> > 2011-06-02 14:12:06.756170 [DEBUG] sofia_glue.c:3041 Set 2833 dtmf
> payload
> > to 101
> >
> >
> >
> > Note that whilst it says its sticking with ptime of 20 (bolded above),
> our
> > vendor in-the-middle sees us as still using ptime=30. Weird. But, on the
> > same switch with different DID, we get the following "good" result:
> >
> >
> >
> > 2011-06-02 14:15:15.135193 [DEBUG] switch_core_state_machine.c:397
> > (sofia/external/8151231234 at xxx.xxx.xxx.xxx:5060) Running State Change
> CS_NEW
> >
> > 2011-06-02 14:15:15.135193 [DEBUG] sofia.c:3210 Channel
> > sofia/external/8151231234@ xxx.xxx.xxx.xxx:5060 entering state
> > [received][100]
> > 2011-06-02 14:15:15.135193 [DEBUG] sofia.c:3217 Remote SDP:
> >
> >
> > v=0
> >
> >
> > o=Acme_UAS 0 1 IN IP4 xxx.xxx.xxx.xxx
> >
> >
> > s=SIP Media Capabilities
> >
> >
> > c=IN IP4 yyy.yyy.yyy.yyy
> >
> >
> > t=0 0
> >
> >
> > m=audio 52052 RTP/AVP 0 18 101
> >
> >
> > a=rtpmap:0 PCMU/8000
> >
> >
> > a=rtpmap:18 G729/8000
> >
> >
> > a=rtpmap:101 telephone-event/8000
> >
> >
> > a=maxptime:20
> >
> >
> >
> >
> >
> > 2011-06-02 14:15:15.135193 [DEBUG] switch_core_state_machine.c:403
> > (sofia/external/8151231234 at xxx.xxx.xxx.xxx.10:5060) State NEW
> >
> > 2011-06-02 14:15:15.135193 [DEBUG] sofia_glue.c:3081 Audio Codec Compare
> > [PCMU:0:8000:20]/[PCMU:0:8000:20]
> >
> > 2011-06-02 14:15:15.135193 [DEBUG] sofia_glue.c:3092 Bah HUMBUG! Sticking
> > with PCMU at 8000h@20i
> >
> > 2011-06-02 14:15:15.135193 [DEBUG] sofia_glue.c:2039 Set Codec
> > sofia/external/8151231234 at xxx.xxx.xxx.xxx:5060 PCMU/8000 20 ms 160
> samples
> >
> > 2011-06-02 14:15:15.135193 [DEBUG] sofia_glue.c:3041 Set 2833 dtmf
> payload
> > to 101
> >
> >
> > Naturally, the "far end" vendor says they think its a "FreeSwitch
> problem".
> > Huh? For my part, I'd just like to FORCE a ptime of 20 in all cases, no
> > matter the codec, for incoming calls, either DID. Tried, but so far, no
> go.
> >
> > A packet capture showed "bad" call coming in with ptime max of 30, we
> > respond that 30 is OK but them re-invite with ptime set to 20, but it
> > doesn't seem to "catch". On the good DID, they proffer ptime max of 30,
> we
> > respond with ptime of 20 and life is good.
> >
> > Sorry to post again on this. I have put in some hours but still same
> results
> > so far. Any suggestions much appreciated.
> >
> > Regards,
> >
> > Mike
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
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From mgende at gendesign.com  Fri Jun  3 00:30:16 2011
From: mgende at gendesign.com (Michael Gende)
Date: Thu, 2 Jun 2011 15:30:16 -0500
Subject: [Freeswitch-users] ptime mismatch confusion
In-Reply-To: <BANLkTin3HpV-7UnCUiiHLoYHSQBFksx4hQ@mail.gmail.com>
References: <BANLkTim-3A4s9G3eLvHTEssg8=xf=-ysCA@mail.gmail.com>
	<BANLkTimd3fyz3S0ATTTSq19gM+Nthdtdzg@mail.gmail.com>
	<BANLkTin3HpV-7UnCUiiHLoYHSQBFksx4hQ@mail.gmail.com>
Message-ID: <BANLkTinUxgFMJjbuMwV=_MbHh09hMk9wvQ@mail.gmail.com>

Tried one last thing and it appears to have worked! Hallelujah!

Left "scrooge" on, and updated, in the vars.xml, the global codec prefs
settings and added @20i to ones we are using.

Still got the "mismatch" message that showed in the "bad" output (prior
post) but the call worked fine! Just did a few to satisfy myself.

Thanks to Michael and Anthony who read my prior posts and had constructive
suggestions.

Regards,

Mike

On Thu, Jun 2, 2011 at 2:57 PM, Michael Gende <mgende at gendesign.com> wrote:

> That's what I'll try. Thanks.
>
> On Thu, Jun 2, 2011 at 2:55 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Thats not really enough info
>> you should have a full pcap of a single call
>>
>> also you should not be using that codec negotiation scrooge unless its
>> a very specific case.
>>
>> get a trace without the scrooge mode and consider reporting it to
>> consulting at freeswitch.org for expedited commercial assistance.
>>
>>
>>
>>
>
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From rgelfand2 at gmail.com  Fri Jun  3 00:33:43 2011
From: rgelfand2 at gmail.com (Roman Gelfand)
Date: Thu, 2 Jun 2011 16:33:43 -0400
Subject: [Freeswitch-users] Troubleshooting FreeSwitch Configuration
Message-ID: <BANLkTi=Y5hGUGdOzxGQXOTzty-m0msutvA@mail.gmail.com>

I am running into trouble trying to register to ITSP.  The log file shows ...

2011-06-01 21:58:10.950068 [WARNING] sofia_reg.c:401 Timeout
Registering 1XXXXXXXXXX-callcentric.com
2011-06-01 21:58:11.957417 [WARNING] sofia_reg.c:425
1XXXXXXXXXX-callcentric.com Failed Registration [0], setting retry to
30 seconds.
2011-06-01 21:58:42.046331 [NOTICE] sofia_reg.c:367 Registering
1XXXXXXXXXX-callcentric.com

When challenged by 407 message from ITSP, I get

05:03:12.857: Missing contact header in REGISTER response.

Does anyone know what is happening here? or pointers how to troubleshoot this?

Thanks in advance


From maciej.aniserowicz at gmail.com  Thu Jun  2 22:28:25 2011
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Thu, 2 Jun 2011 11:28:25 -0700 (PDT)
Subject: [Freeswitch-users] Eavesdrop between 2 FS instaces
Message-ID: <1307039305975-6432366.post@n2.nabble.com>

Hello, 
Is it possible to eavesdrop between 2 FS instaces? I have CallA parked on
FS1 and want to eavesdrop on CallB that is on FS2. I have both channel ids. 

I tried transfering CallA to FS2, intercept CallB in FS2 dialplan and then
eavesdrop in inline dialplan, but this does not work (however it works with
bridging). Can I make it work somehow? 

Best regards, 
Maciej Aniserowicz

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Eavesdrop-between-2-FS-instaces-tp6432366p6432366.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From joaocarlosleme at gmail.com  Fri Jun  3 01:02:35 2011
From: joaocarlosleme at gmail.com (Joao Leme)
Date: Thu, 2 Jun 2011 14:02:35 -0700
Subject: [Freeswitch-users] FIFO suddenly stopped working properly
Message-ID: <BANLkTim8MJUDxc+0aQusxckV4+X6mrHwDA@mail.gmail.com>

Hi there,
No change nor updates have been made in months. Everything was working great
but now FIFO is not ringing the extensions while the first call is not HUNG
UP...meaning, first call to go to FIFO rings the agents, and one
person answer...when the second call comes in, it goes to fifo queue (caller
listens the music) but no one is aware of the call, nor even the agents that
are available. Only after the first call hangs up the 2nd goes in. I tried
troubleshooting but no help.

What I also noticed is that before, to display the caller ID on my sip
client I had to use:

<action application="set"
data="origination_caller_id_number=${caller_id_number}"/>
<action application="set"
data="origination_caller_id_name=${caller_id_name}"/>

but now, while testing, the caller id was showing when calling
fifo without setting those parameters, but before it would just show the
fifo queue name.

All I can say is that I can't figure out why and how to fix these behavior
if I have done no change to FreeSWITCH nor the config files. I also tried
downloading the latest GIT and got the same response.

THANKS,
John
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From david.ponzone at ipeva.fr  Fri Jun  3 01:21:14 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 2 Jun 2011 23:21:14 +0200
Subject: [Freeswitch-users] Hanged up callers doubt
In-Reply-To: <BANLkTikV_-=Zkv8zOQVcWyg5Gc06UtRxjw@mail.gmail.com>
References: <BANLkTi=qf4T3Wb9_2TmJdt4Wwif5qipsmg@mail.gmail.com>
	<BANLkTik+DYhPT1ss8-OqpQsvUAA30A5L=Q@mail.gmail.com>
	<BANLkTinxTu=c_9usFNv0sOOWdQ3-DnAYXQ@mail.gmail.com>
	<BANLkTi=g3J1nZTKArouGSJMDHVSE1L0idA@mail.gmail.com>
	<BANLkTi=WN97QMJcpwyEGQNvRxBW1c=UsuQ@mail.gmail.com>
	<BANLkTiky4M2ZgTMS228-DJ6oaEsmkpG19g@mail.gmail.com>
	<BANLkTimzsNxVyFpLf6x8X4E7P58FYwdHng@mail.gmail.com>
	<BANLkTinYynZsX6=zg-yyeP7hEQ7-q5J9+Q@mail.gmail.com>
	<BANLkTim=ueSmFA_ug2MbbLFMBh0362_2Sg@mail.gmail.com>
	<BANLkTikV_-=Zkv8zOQVcWyg5Gc06UtRxjw@mail.gmail.com>
Message-ID: <2C18EE80-BB65-4A93-9A44-627B6154DD5F@ipeva.fr>

Sure, that one is easy:

from fs_cli:

conference <confname> kick all

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

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

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




Le 02/06/2011 ? 21:33, Leonardo P. Bidinoto a ?crit :

> Hi,
> 
> About this issue, could be a way to destroy the conference from FS, in a attempt to remove the stucked channels from the "show channels " and "conference list"?
> 
> 2011/5/17 Leonardo P. Bidinoto <leonardo.bidinoto at voicetechnology.com.br>
> Sure. Im sending a pcap file made by tcpdump and one that i made by ngrep. In both files, it was registering whats happening when i stuck the channel by hanging up while using a ESL connection inside a conference(app socket 8085 sync full). I did a "conference kick" command in this channel while its was waiting to close the ESL connection.
> 
> 
> 
> 2011/5/16 Michael Collins <msc at freeswitch.org>
> Can you tcpdump or otherwise capture the traffic on port 8085? I am curious what is happening with that.
> -MC
> 
> 
> On Mon, May 16, 2011 at 12:12 PM, Leonardo P. Bidinoto <leonardo.bidinoto at voicetechnology.com.br> wrote:
> hehe, ok michael.
> 
> here is the pastebin link:
> http://pastebin.freeswitch.org/16303
> 
> 
> 2011/5/13 Michael Collins <msc at freeswitch.org>
> Pastebin this info and select "FreeSWITCH Log" as the syntax highlighting. I need the colorized output to read logs. (I'm getting older and it's hard for me to ready black and white in an email.)
> 
> -MC
> 
> 
> On Fri, May 13, 2011 at 7:32 AM, Leonardo P. Bidinoto <leonardo.bidinoto at voicetechnology.com.br> wrote:
> Hi Michael,
> 
> Just succeeded to reproduce the problem.
> 
> The condition is: when a channel inside a conference is using a ESL connection(lets call it "A") through socket application and another ESL connection(lets call it "B") executes a command with this channel, the "B" ESL connection will wait the "A" ESL connection close to execute its command.
> If the channel hangs up before the "A" ESL connection is closed, then "B" ESL command will never be executed and the stucked channel will still be there, into sofia and the conference too.
> To verify that, just do "show channels" and "conference list". with "uuid_exists" command, return "false".
> 
> Here are the actions done by the channel before get stucked:
> 
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.187321 [NOTICE] switch_channel.c:816 New Channel sofia/external/1000123402 at 192.168.0.154 [16e09413-9cb0-4011-a635-f91933a35c0f]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154 entering state [received][100]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] sofia.c:4772 Remote SDP:
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] sofia_glue.c:4656 Audio Codec Compare [BV32:107:16000:20:0]/[PCMU:0:8000:20:64000]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] sofia_glue.c:4656 Audio Codec Compare [BV32:107:16000:20:0]/[PCMA:8:8000:20:64000]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] sofia_glue.c:4656 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] sofia_glue.c:2788 Set Codec sofia/external/1000123402 at 192.168.0.154 PCMU/8000 20 ms 160 samples 64000 bits
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] sofia_glue.c:4770 Set 2833 dtmf send/recv payload to 101
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] sofia.c:4943 (sofia/external/1000123402 at 192.168.0.154) State Change CS_NEW -> CS_INIT
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/1000123402 at 192.168.0.154 [BREAK]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154) Running State Change CS_INIT
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] switch_core_state_machine.c:361 (sofia/external/1000123402 at 192.168.0.154) State INIT
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] mod_sofia.c:84 sofia/external/1000123402 at 192.168.0.154 SOFIA INIT
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] mod_sofia.c:124 (sofia/external/1000123402 at 192.168.0.154) State Change CS_INIT -> CS_ROUTING
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/1000123402 at 192.168.0.154 [BREAK]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] switch_core_state_machine.c:361 (sofia/external/1000123402 at 192.168.0.154) State INIT going to sleep
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154) Running State Change CS_ROUTING
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] switch_channel.c:1665 (sofia/external/1000123402 at 192.168.0.154) Callstate Change DOWN -> RINGING
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] switch_core_state_machine.c:364 (sofia/external/1000123402 at 192.168.0.154) State ROUTING
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] mod_sofia.c:147 sofia/external/1000123402 at 192.168.0.154 SOFIA ROUTING
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] switch_core_state_machine.c:77 sofia/external/1000123402 at 192.168.0.154 Standard ROUTING
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [INFO] mod_dialplan_xml.c:331 Processing Phone1 <1000123402>->1234567890 in context public
> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/1000123402 at 192.168.0.154 parsing [public->unloop] continue=false
> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/1000123402 at 192.168.0.154 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/1000123402 at 192.168.0.154 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/1000123402 at 192.168.0.154 parsing [public->public_extensions] continue=false
> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/1000123402 at 192.168.0.154 Regex (PASS) [public_extensions] destination_number(1234567890) =~ /^(\d*)$/ break=on-false
> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/1000123402 at 192.168.0.154 Action transfer(1234567890 XML default) 
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG] switch_core_state_machine.c:119 (sofia/external/1000123402 at 192.168.0.154) State Change CS_ROUTING -> CS_EXECUTE
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/1000123402 at 192.168.0.154 [BREAK]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG] switch_core_state_machine.c:364 (sofia/external/1000123402 at 192.168.0.154) State ROUTING going to sleep
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG] switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154) Running State Change CS_EXECUTE
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG] switch_core_state_machine.c:371 (sofia/external/1000123402 at 192.168.0.154) State EXECUTE
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG] mod_sofia.c:240 sofia/external/1000123402 at 192.168.0.154 SOFIA EXECUTE
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG] switch_core_state_machine.c:157 sofia/external/1000123402 at 192.168.0.154 Standard EXECUTE
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 transfer(1234567890 XML default)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG] switch_ivr.c:1597 (sofia/external/1000123402 at 192.168.0.154) State Change CS_EXECUTE -> CS_ROUTING
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/1000123402 at 192.168.0.154 [BREAK]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG] switch_core_session.c:707 Send signal sofia/external/1000123402 at 192.168.0.154 [BREAK]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [NOTICE] switch_ivr.c:1603 Transfer sofia/external/1000123402 at 192.168.0.154 to XML[1234567890 at default]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG] switch_core_state_machine.c:371 (sofia/external/1000123402 at 192.168.0.154) State EXECUTE going to sleep
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG] switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154) Running State Change CS_ROUTING
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG] switch_core_state_machine.c:364 (sofia/external/1000123402 at 192.168.0.154) State ROUTING
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG] mod_sofia.c:147 sofia/external/1000123402 at 192.168.0.154 SOFIA ROUTING
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG] switch_core_state_machine.c:77 sofia/external/1000123402 at 192.168.0.154 Standard ROUTING
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [INFO] mod_dialplan_xml.c:331 Processing Phone1 <1000123402>->1234567890 in context default
> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/1000123402 at 192.168.0.154 parsing [default->flex] continue=false
> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/1000123402 at 192.168.0.154 Regex (PASS) [flex] destination_number(1234567890) =~ /^(\d+)$/ break=on-false
> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/1000123402 at 192.168.0.154 Action log(INFO VOICE received dest=1234567890) 
> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/1000123402 at 192.168.0.154 Action set(playback_terminators=#) 
> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/1000123402 at 192.168.0.154 Action log(INFO Let's do some ivrd, shall we?) 
> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/1000123402 at 192.168.0.154 Action set(ivr_path=/usr/local/ipconf/fs/daemon/ivr.rb) 
> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/1000123402 at 192.168.0.154 Action socket(localhost:8084 full) 
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG] switch_core_state_machine.c:119 (sofia/external/1000123402 at 192.168.0.154) State Change CS_ROUTING -> CS_EXECUTE
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/1000123402 at 192.168.0.154 [BREAK]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG] switch_core_state_machine.c:364 (sofia/external/1000123402 at 192.168.0.154) State ROUTING going to sleep
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG] switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154) Running State Change CS_EXECUTE
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG] switch_core_state_machine.c:371 (sofia/external/1000123402 at 192.168.0.154) State EXECUTE
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG] mod_sofia.c:240 sofia/external/1000123402 at 192.168.0.154 SOFIA EXECUTE
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG] switch_core_state_machine.c:157 sofia/external/1000123402 at 192.168.0.154 Standard EXECUTE
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 log(INFO VOICE received dest=1234567890)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [INFO] mod_dptools.c:1184 VOICE received dest=1234567890
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 set(playback_terminators=#)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331 [DEBUG] mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET [playback_terminators]=[#]
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 log(INFO Let's do some ivrd, shall we?)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331 [INFO] mod_dptools.c:1184 Let's do some ivrd, shall we?
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 set(ivr_path=/usr/local/ipconf/fs/daemon/ivr.rb)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331 [DEBUG] mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET [ivr_path]=[/usr/local/ipconf/fs/daemon/ivr.rb]
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 socket(localhost:8084 full)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute answer()
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 answer()
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309 [DEBUG] sofia_glue.c:3022 AUDIO RTP [sofia/external/1000123402 at 192.168.0.154] 192.168.0.154 port 24232 -> 192.168.0.111 port 4046 codec: 0 ms: 20
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309 [DEBUG] switch_rtp.c:1623 Starting timer [soft] 160 bytes per 20ms
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG] sofia_glue.c:3284 Set 2833 dtmf send payload to 101
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG] sofia_glue.c:3289 Set 2833 dtmf receive payload to 101
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG] mod_sofia.c:681 Local SDP sofia/external/1000123402 at 192.168.0.154:
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG] switch_core_session.c:707 Send signal sofia/external/1000123402 at 192.168.0.154 [BREAK]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG] switch_channel.c:2827 (sofia/external/1000123402 at 192.168.0.154) Callstate Change RINGING -> ACTIVE
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [NOTICE] mod_dptools.c:930 Channel [sofia/external/1000123402 at 192.168.0.154] has been answered
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.933312 [DEBUG] sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154 entering state [completed][200]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.040285 [DEBUG] sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154 entering state [ready][200]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.082294 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute playback(/usr/local/freeswitch/sounds/flex/app32/teste/WelcomeToTheConferenceSystem.wav)
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 playback(/usr/local/freeswitch/sounds/flex/app32/teste/WelcomeToTheConferenceSystem.wav)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.082294 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.114277 [DEBUG] switch_ivr_play_say.c:1649 done playing file
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.155281 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Mainmenu.wav flex_digits 5000 )
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Mainmenu.wav flex_digits 5000 )
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.155281 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.893670 [DEBUG] switch_rtp.c:3280 RTP RECV DTMF 1:1120
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.893670 [DEBUG] switch_ivr_play_say.c:1649 done playing file
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.897674 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute set(flex_digits)
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 set(flex_digits)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.897674 [DEBUG] mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET [flex_digits]=[UNDEF]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.940673 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/PleaseEnterYourPassCodeNow.wav flex_digits 5000 )
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/PleaseEnterYourPassCodeNow.wav flex_digits 5000 )
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.940673 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.113662 [DEBUG] switch_rtp.c:3280 RTP RECV DTMF 8:640
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.113662 [DEBUG] switch_ivr_play_say.c:1649 done playing file
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.116663 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute set(flex_digits)
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 set(flex_digits)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.116663 [DEBUG] mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET [flex_digits]=[UNDEF]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.117663 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute read(1 11 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 5000 #,*)
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 read(1 11 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 5000 #,*)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.117663 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.153658 [DEBUG] switch_ivr_play_say.c:1649 done playing file
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.053615 [DEBUG] switch_rtp.c:3280 RTP RECV DTMF #:960
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.055618 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute set(flex_digits)
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 set(flex_digits)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.055618 [DEBUG] mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET [flex_digits]=[UNDEF]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.146613 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute playback(/usr/local/freeswitch/sounds/flex/app32/teste/InternalHelp.wav)
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 playback(/usr/local/freeswitch/sounds/flex/app32/teste/InternalHelp.wav)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.146613 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603 [DEBUG] switch_rtp.c:3280 RTP RECV DTMF #:800
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603 [DEBUG] mod_dptools.c:1664 Digit #
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603 [DEBUG] switch_ivr_play_say.c:1649 done playing file
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.350617 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute playback(/usr/local/freeswitch/sounds/flex/app32/teste/YouAreTheOnlyPersonOnThisConference.wav)
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 playback(/usr/local/freeswitch/sounds/flex/app32/teste/YouAreTheOnlyPersonOnThisConference.wav)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.350617 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.253520 [DEBUG] switch_ivr_play_say.c:1649 done playing file
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.322510 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute conference(15646 at teste+flags{waste})
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 conference(15646 at teste+flags{waste})
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492 [DEBUG] mod_conference.c:5582 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492 [DEBUG] mod_conference.c:5627 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492 [DEBUG] switch_core_codec.c:116 sofia/external/1000123402 at 192.168.0.154 Push codec L16:70
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:05.280469 [DEBUG] switch_core_session.c:707 Send signal sofia/external/1000123402 at 192.168.0.154 [BREAK]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:05.280469 [DEBUG] mod_conference.c:2557 Setup timer soft success interval: 20  samples: 160
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.573928 [DEBUG] switch_rtp.c:3280 RTP RECV DTMF *:960
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.593929 [DEBUG]  mod_conference.c:2021 Execute app: socket, localhost:8085 sync full
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.593929 [DEBUG] switch_core_codec.c:141 sofia/external/1000123402 at 192.168.0.154 Restore previous codec PCMU:0.
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 socket(localhost:8085 sync full)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.097927 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 )
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 )
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.097927 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.133923 [DEBUG] switch_ivr_play_say.c:1649 done playing file
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.373913 [DEBUG] switch_rtp.c:3280 RTP RECV DTMF 1:1120
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.375912 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute set(flex_digits)
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 set(flex_digits)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.375912 [DEBUG] mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET [flex_digits]=[UNDEF]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.490911 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute playback(/usr/local/freeswitch/sounds/flex/app32/teste/SelfMuteOn.wav)
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 playback(/usr/local/freeswitch/sounds/flex/app32/teste/SelfMuteOn.wav)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.490911 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.233824 [DEBUG] switch_ivr_play_say.c:1649 done playing file
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.235828 [DEBUG] switch_cpp.cpp:988 sofia/external/1000123402 at 192.168.0.154 destroy/unlink session from object
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.235828 [DEBUG] switch_core_codec.c:116 sofia/external/1000123402 at 192.168.0.154 Push codec L16:70
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.433668 [DEBUG] switch_rtp.c:3280 RTP RECV DTMF *:800
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.453667 [DEBUG]  mod_conference.c:2021 Execute app: socket, localhost:8085 sync full
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.453667 [DEBUG] switch_core_codec.c:141 sofia/external/1000123402 at 192.168.0.154 Restore previous codec PCMU:0.
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 socket(localhost:8085 sync full)
> ====================================================================================================================================================
> While Inside this connection, a "conference 15646 kick [member_id of this channels]" command is executed by a fs_cli console and get stuck while waiting response.
> ====================================================================================================================================================
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.000850 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 )
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 )
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.000850 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.033846 [DEBUG] switch_ivr_play_say.c:1649 done playing file
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.553821 [DEBUG] switch_rtp.c:3280 RTP RECV DTMF 1:960
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.555825 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute set(flex_digits)
> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 set(flex_digits)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.555825 [DEBUG] mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET [flex_digits]=[UNDEF]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [DEBUG] switch_channel.c:2560 (sofia/external/1000123402 at 192.168.0.154) Callstate Change ACTIVE -> HANGUP
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [NOTICE] sofia.c:538 Hangup sofia/external/1000123402 at 192.168.0.154 [CS_EXECUTE] [NORMAL_CLEARING]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [DEBUG] switch_channel.c:2576 Send signal sofia/external/1000123402 at 192.168.0.154 [KILL]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/1000123402 at 192.168.0.154 [BREAK]
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG] switch_core_session.c:2057 sofia/external/1000123402 at 192.168.0.154 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG] switch_cpp.cpp:988 sofia/external/1000123402 at 192.168.0.154 destroy/unlink session from object
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG] switch_core_session.c:2057 sofia/external/1000123402 at 192.168.0.154 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG] switch_core_codec.c:116 sofia/external/1000123402 at 192.168.0.154 Push codec L16:70
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG] mod_conference.c:2815 Channel leaving conference, cause: NORMAL_CLEARING
> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.757239 [DEBUG] mod_conference.c:6104 sofia/external/1000123402 at 192.168.0.154 skip receive message [UNBRIDGE] (channel is hungup already)
> 
> I hope this info helps.
> 
> 2011/5/12 Michael Collins <msc at freeswitch.org>
> 
> 
> On Thu, May 12, 2011 at 12:27 PM, Leonardo P. Bidinoto <leonardo.bidinoto at voicetechnology.com.br> wrote:
> Hi Michael,
> 
> Im not using to any cdr module.
> 
> I would recommend that you do several things:
> 
> #1 - update to latest git
> #2 - rotate logs
> #3 - enable uuid logging (see logfile.conf.xml - param name "uuid")
> #4 - reproduce the symptom with a single call (if possible)
> #5 - pastebin the log for the uuid in question and link to it in this thread
> 
> From there hopefully we'll get a clue as to what is happening.
> -MC
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Leonardo Pires Bidinoto
> Voice Technology
> www.voicetechnology.com.br
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Leonardo Pires Bidinoto
> Voice Technology
> www.voicetechnology.com.br
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Leonardo Pires Bidinoto
> Voice Technology
> www.voicetechnology.com.br
> 
> 
> 
> -- 
> Leonardo Pires Bidinoto
> Voice Technology
> www.voicetechnology.com.br
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From steveayre at gmail.com  Fri Jun  3 01:54:31 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 2 Jun 2011 22:54:31 +0100
Subject: [Freeswitch-users] Eavesdrop between 2 FS instaces
In-Reply-To: <1307039305975-6432366.post@n2.nabble.com>
References: <1307039305975-6432366.post@n2.nabble.com>
Message-ID: <BANLkTinLdaHeaMy9XCthLM87jxQKNGSBXw@mail.gmail.com>

You can't eavesdrop on a call that's going through another server, because
you're not in the media path.

As a possible workaround, it would be possible to dial into FS2 from FS1
into a special context (which can be done either by using a different
profile for this, or setting the context in the user directory for a special
fs-to-fs user) and do the eavesdrop on FS2 that way. You could use the
sip-X_* variables to send the UUID to eavesdrop on to FS2 in the INVITE from
FS1.

-Steve




On 2 June 2011 19:28, Maciej Aniserowicz <maciej.aniserowicz at gmail.com>wrote:

> Hello,
> Is it possible to eavesdrop between 2 FS instaces? I have CallA parked on
> FS1 and want to eavesdrop on CallB that is on FS2. I have both channel ids.
>
> I tried transfering CallA to FS2, intercept CallB in FS2 dialplan and then
> eavesdrop in inline dialplan, but this does not work (however it works with
> bridging). Can I make it work somehow?
>
> Best regards,
> Maciej Aniserowicz
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Eavesdrop-between-2-FS-instaces-tp6432366p6432366.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Fri Jun  3 01:57:12 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 2 Jun 2011 22:57:12 +0100
Subject: [Freeswitch-users] Troubleshooting FreeSwitch Configuration
In-Reply-To: <BANLkTi=Y5hGUGdOzxGQXOTzty-m0msutvA@mail.gmail.com>
References: <BANLkTi=Y5hGUGdOzxGQXOTzty-m0msutvA@mail.gmail.com>
Message-ID: <BANLkTimZ4Ck7rfFeRRgxSu0-U_0OG19PcA@mail.gmail.com>

Sounds like they're not sending back a valid response.

It would be useful to see what's beeing sent back and forth. Change the
logging level to debug and do 'sofia global siptrace on' to log all sip
packets so we can see what's in them. It could be a problem with their
server not sending the correct reply or it could be that your configuration
isn't quite right to interop with their server and needs tweaking.

-Steve


On 2 June 2011 21:33, Roman Gelfand <rgelfand2 at gmail.com> wrote:

> I am running into trouble trying to register to ITSP.  The log file shows
> ...
>
> 2011-06-01 21:58:10.950068 [WARNING] sofia_reg.c:401 Timeout
> Registering 1XXXXXXXXXX-callcentric.com
> 2011-06-01 21:58:11.957417 [WARNING] sofia_reg.c:425
> 1XXXXXXXXXX-callcentric.com Failed Registration [0], setting retry to
> 30 seconds.
> 2011-06-01 21:58:42.046331 [NOTICE] sofia_reg.c:367 Registering
> 1XXXXXXXXXX-callcentric.com
>
> When challenged by 407 message from ITSP, I get
>
> 05:03:12.857: Missing contact header in REGISTER response.
>
> Does anyone know what is happening here? or pointers how to troubleshoot
> this?
>
> Thanks in advance
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jan.berger at video24.no  Fri Jun  3 04:07:03 2011
From: jan.berger at video24.no (Jan Berger)
Date: Fri, 3 Jun 2011 02:07:03 +0200
Subject: [Freeswitch-users] mrcp
In-Reply-To: <BANLkTi=Q6oMRUNq9GSVjMORwxS0eWV2=9w@mail.gmail.com>
References: <A358176020C045358C04C08944501FAA@dell9400>
	<BANLkTi=Q6oMRUNq9GSVjMORwxS0eWV2=9w@mail.gmail.com>
Message-ID: <D663E566A50142F69915A5AFB17257F7@dell9400>

Thanks,

 

It "kind of" does the job.It's nice to be able to specify URL or file as
grammar, but I need to pass SRGS or SSML directly.

 

If you attempt to send SRGS it screws up in mod_unimrcp and attempt to load
a file from what I can see - it seems to lack a test to see if it gets
passed srgs or ssml from the app directly. Need to test tomorrow.

 

Jan

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven
Ayre
Sent: 2. juni 2011 13:44
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mrcp

 

http://wiki.freeswitch.org/wiki/Mod_unimrcp

:)




On 2 June 2011 11:25, Jan Berger <jan.berger at video24.no> wrote:

Hi,

 

I would like to communicate with MRCP to control TTS and ASR from a C module
- how can I do that easiest?

 

Jan

 


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

 

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From rgelfand2 at gmail.com  Fri Jun  3 04:33:36 2011
From: rgelfand2 at gmail.com (Roman Gelfand)
Date: Thu, 2 Jun 2011 20:33:36 -0400
Subject: [Freeswitch-users] Troubleshooting FreeSwitch Configuration
In-Reply-To: <BANLkTimZ4Ck7rfFeRRgxSu0-U_0OG19PcA@mail.gmail.com>
References: <BANLkTi=Y5hGUGdOzxGQXOTzty-m0msutvA@mail.gmail.com>
	<BANLkTimZ4Ck7rfFeRRgxSu0-U_0OG19PcA@mail.gmail.com>
Message-ID: <BANLkTi=Qi2R5kXSUcy1Y4i8D++W5q4YzHA@mail.gmail.com>

Thanks for your help.

The actual application I am using is Karoo Bridge SBC which is built
on FreeSwitch.  I found freeswitch.xml.fsxml configuration file.  I
suppose this the final resolved configuraiton.  I am not sure where I
set debug option.

Anyway, I thought the following could be useful for you to see.


09:03:38.274:   [CID=29f64488] B2B Transaction CREATED -
register13166648z9hG4bKr84Dr2gK3ccHm
09:03:38.274:   [CID=29f64488] JS: Setting transaction property
respond-to-packet-source=1
09:03:38.274:   [CID=29f64488] JS: Setting transaction property
auth-method=access-list
09:03:38.274:   [CID=29f64488] JS: Setting transaction property auth-method=none
09:03:38.310:   [CID=29f64488] >>> REGISTER
sip:callcentric.com;transport=udp SIP/2.0 LEN: 843 SRC:
XX.XX.XXX.XXX:5060 DST: 204.11.192.37:5060 ENC: 0 PROT: udp
09:03:38.310: ^M
{^M
                [CID=29f64488] REGISTER
sip:callcentric.com;transport=udp SIP/2.0^M
                [CID=29f64488] From:
<sip:1##########@callcentric.com;transport=udp>;tag=v0r8KQHmF054m^M
                [CID=29f64488] To:
<sip:1##########@callcentric.com;transport=udp>^M
                [CID=29f64488] Call-ID: 99b2a5a7-1e7c-4e65-b9ac-6fadbf4ca1c3^M
                [CID=29f64488] CSeq: 13166648 REGISTER^M
                [CID=29f64488] Expires: 1800^M
                [CID=29f64488] User-Agent: OSS Karoo FS^M
                [CID=29f64488] Allow: INVITE, ACK, BYE, CANCEL,
OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY^M
                [CID=29f64488] Supported: timer, precondition, path, replaces^M
                [CID=29f64488] Proxy-Authorization: Digest
username="1##########", realm="callcentric.com",
nonce="a0667bbc4a83b98f36b9619405daa1f5", opaque="", algorithm=MD5,
uri="sip:XX.XX.XXX.XXX:5060",
response="d4ee3820ca6f33ac8d40afc319b82006"^M
                [CID=29f64488] Content-Length: 0^M
                [CID=29f64488] Max-Forwards: 69^M
                [CID=29f64488] Contact:
<sip:1##########@XX.XX.XXX.XXX:5060;transport=udp>^M
                [CID=29f64488] Via: SIP/2.0/UDP
XX.XX.XXX.XXX:5060;branch=z9hG4bK34487872-8cf7-11e0-a33e-0800274986dc;rport^M
};
09:03:38.279:   [CID=29f64488] B2B Transaction DESTROYED -
register13166647z9hG4bKQZBNp7ZF63Nyr
09:03:38.348:   [CID=29f64488] Found Transaction
register13166648z9hG4bK34487872-8cf7-11e0-a33e-0800274986dc
09:03:38.348:   [CID=29f64488] <<< SIP/2.0 407 Proxy Authentication
Required LEN: 498 SRC: 204.11.192.37:5060 DST: XX.XX.XXX.XXX:5060 EXT:
[XX.XX.XXX.XXX] ENC: 0 PROT: udp
09:03:38.348: ^M
{^M
                [CID=29f64488] SIP/2.0 407 Proxy Authentication Required^M
                [CID=29f64488] v: SIP/2.0/UDP
XX.XX.XXX.XXX:5060;branch=z9hG4bK34487872-8cf7-11e0-a33e-0800274986dc;rport=5060^M
                [CID=29f64488] f:
<sip:1##########@callcentric.com;transport=udp>;tag=v0r8KQHmF054m^M
                [CID=29f64488] t:
<sip:1##########@callcentric.com;transport=udp>^M
                [CID=29f64488] i: 99b2a5a7-1e7c-4e65-b9ac-6fadbf4ca1c3^M
                [CID=29f64488] CSeq: 13166648 REGISTER^M
                [CID=29f64488] Proxy-Authenticate: Digest
realm="callcentric.com", domain="sip:callcentric.com",
nonce="edbfefbbc2ab6d684bae2b1de3fec2ea", opaque="", stale=TRUE,
algorithm=MD5^M
                [CID=29f64488] l: 0^M
};
09:03:38.349: Missing contact header in REGISTER response.
09:03:38.350:   [CID=29f64488] >>> SIP/2.0 407 Proxy Authentication
Required LEN: 526 SRC: XX.XX.XXX.XXX:5060 DST: XX.XX.XXX.XXX:5080 ENC:
0 PROT: udp
09:03:38.351: ^M
{^M
                [CID=29f64488] SIP/2.0 407 Proxy Authentication Required^M
                [CID=29f64488] f:
<sip:1##########@callcentric.com;transport=udp>;tag=v0r8KQHmF054m^M
                [CID=29f64488] t:
<sip:1##########@callcentric.com;transport=udp>^M
                [CID=29f64488] i: 99b2a5a7-1e7c-4e65-b9ac-6fadbf4ca1c3^M
                [CID=29f64488] CSeq: 13166648 REGISTER^M
                [CID=29f64488] Proxy-Authenticate: Digest
realm="callcentric.com", domain="sip:callcentric.com",
nonce="edbfefbbc2ab6d684bae2b1de3fec2ea", opaque="", stale=TRUE,
algorithm=MD5^M
                [CID=29f64488] l: 0^M





On Thu, Jun 2, 2011 at 5:57 PM, Steven Ayre <steveayre at gmail.com> wrote:
> Sounds like they're not sending back a valid response.
>
> It would be useful to see what's beeing sent back and forth. Change the
> logging level to debug and do 'sofia global siptrace on' to log all sip
> packets so we can see what's in them. It could be a problem with their
> server not sending the correct reply or it could be that your configuration
> isn't quite right to interop with their server and needs tweaking.
>
> -Steve
>
>
> On 2 June 2011 21:33, Roman Gelfand <rgelfand2 at gmail.com> wrote:
>>
>> I am running into trouble trying to register to ITSP. ?The log file shows
>> ...
>>
>> 2011-06-01 21:58:10.950068 [WARNING] sofia_reg.c:401 Timeout
>> Registering 1XXXXXXXXXX-callcentric.com
>> 2011-06-01 21:58:11.957417 [WARNING] sofia_reg.c:425
>> 1XXXXXXXXXX-callcentric.com Failed Registration [0], setting retry to
>> 30 seconds.
>> 2011-06-01 21:58:42.046331 [NOTICE] sofia_reg.c:367 Registering
>> 1XXXXXXXXXX-callcentric.com
>>
>> When challenged by 407 message from ITSP, I get
>>
>> 05:03:12.857: Missing contact header in REGISTER response.
>>
>> Does anyone know what is happening here? or pointers how to troubleshoot
>> this?
>>
>> Thanks in advance
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From cmcureau at gmail.com  Fri Jun  3 05:39:22 2011
From: cmcureau at gmail.com (Chris Cureau)
Date: Thu, 2 Jun 2011 20:39:22 -0500
Subject: [Freeswitch-users] Call For Help: FreeSWITCH Cookbook
In-Reply-To: <BANLkTinuuqCh4duDNsnkt33k1aNgqmAZuw@mail.gmail.com>
References: <BANLkTinuuqCh4duDNsnkt33k1aNgqmAZuw@mail.gmail.com>
Message-ID: <332535808673479297@unknownmsgid>

--
Chris Cureau
Director of Information Services
R&R Home Care, Inc.
(985) 624-3800

On May 31, 2011, at 12:37 PM, Michael Collins <msc at freeswitch.org> wrote:

> Hello FreeSWITCHers!
>
> We have a need for some assistance with a few of our recipes for the cookbook. If you are able to write technical documentation in English and have first-hand knowledge of these topics then you might be in a position to help:
>
> CDRs:
> Parsing XML CDRs
> Handling A and B leg CDRs
> Using a web server to handle XML CDRs
>
> Event Socket:
> Inbound ESL connections (basic how-to)
> Outbound ESL connections (basic how-to)
> Launch an outbound call with inbound event socket & ESL
> Handle inbound call with socket app & ESL
> Reacting to events (events for "my call" vs. system-wide events, etc.)
>
> Misc:
> Presence for BLF and SLA
> Presence for FIFO agent status
>
> If you think you've got the chops to write one or more recipes for the cookbook then email me off list and we'll discuss the specifics. I will get you all the information you need to get started.
>
> Thanks!
>
> -Michael
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From cmrienzo at gmail.com  Fri Jun  3 06:28:27 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Thu, 2 Jun 2011 22:28:27 -0400
Subject: [Freeswitch-users] mrcp
In-Reply-To: <D663E566A50142F69915A5AFB17257F7@dell9400>
References: <A358176020C045358C04C08944501FAA@dell9400>
	<BANLkTi=Q6oMRUNq9GSVjMORwxS0eWV2=9w@mail.gmail.com>
	<D663E566A50142F69915A5AFB17257F7@dell9400>
Message-ID: <BANLkTim7qr0qbsxcjadrshjEfka_kZ+SYA@mail.gmail.com>

To send SRGS directly, use the "inline:" prefix.   For TTS, you can send
SSML if your server supports it.

This is the APP to control speech recognition:

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_detect_speech

I need to update the wiki... it got out of sync last year when luke-jr added
support for multiple grammars per session.

Chris


On Thu, Jun 2, 2011 at 8:07 PM, Jan Berger <jan.berger at video24.no> wrote:

>  Thanks,
>
>
>
> It ?kind of? does the job.It?s nice to be able to specify URL or file as
> grammar, but I need to pass SRGS or SSML directly.
>
>
>
> If you attempt to send SRGS it screws up in mod_unimrcp and attempt to
> load a file from what I can see ? it seems to lack a test to see if it gets
> passed srgs or ssml from the app directly. Need to test tomorrow.
>
>
>
> Jan
>  ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* 2. juni 2011 13:44
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] mrcp
>
>
>
> http://wiki.freeswitch.org/wiki/Mod_unimrcp
>
> :)
>
>
>  On 2 June 2011 11:25, Jan Berger <jan.berger at video24.no> wrote:
>
> Hi,
>
>
>
> I would like to communicate with MRCP to control TTS and ASR from a C
> module - how can I do that easiest?
>
>
>
> Jan
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Fri Jun  3 08:39:04 2011
From: u2nsam at gmail.com (Sam)
Date: Fri, 3 Jun 2011 10:09:04 +0530
Subject: [Freeswitch-users] manipulate CID
In-Reply-To: <BANLkTi=ONknfmsocwr+4sQTK54RYUSCTcQ@mail.gmail.com>
References: <BANLkTi=ONknfmsocwr+4sQTK54RYUSCTcQ@mail.gmail.com>
Message-ID: <BANLkTi=mDgriNWGn=7cfXXAKzwOR6EZpqA@mail.gmail.com>

I still cannot get desired caller id through prefixing,

i get from header as :-  From: "" <sip:91 at xxx.xxx.xxx.xxx>


<extension name="novanet_7001">
            <condition field="destination_number" expression="^(7001)"
continue="false">
            <action application="set" data="call_timeout=20"/>
            <action application="info" />
            <action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED"/>

            <action application="set"
data="effective_caller_id_number=91${Caller-Caller-ID-Number}"/>
            <action application="ring_ready" />
            <action application="bridge"
data="{sip_cid_type=rpid}sofia/gateway/novanet/1646xxxxxxxx"/>

</condition>
        </extension>

Logs:
-----------------------

Dialplan: FreeTDM/1:13/7001 Action
set(continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED)
Dialplan: FreeTDM/1:13/7001 Action
set(effective_caller_id_number=91${Caller-Caller-ID-Number})
Dialplan: FreeTDM/1:13/7001 Action ring_ready()
Dialplan: FreeTDM/1:13/7001 Action
bridge({sip_cid_type=rpid}sofia/gateway/novanet/1646xxxxxxx)
2011-06-03 03:52:34.765627 [DEBUG] switch_core_state_machine.c:119
(FreeTDM/1:13/7001) State Change CS_ROUTING -> CS_EXECUTE
2011-06-03 03:52:34.765627 [DEBUG] switch_core_session.c:1116 Send signal
FreeTDM/1:13/7001 [BREAK]
2011-06-03 03:52:34.765627 [DEBUG] switch_core_state_machine.c:359
(FreeTDM/1:13/7001) State ROUTING going to sleep
2011-06-03 03:52:34.765627 [DEBUG] switch_core_state_machine.c:320
(FreeTDM/1:13/7001) Running State Change CS_EXECUTE
2011-06-03 03:52:34.765627 [DEBUG] switch_core_state_machine.c:366
(FreeTDM/1:13/7001) State EXECUTE
2011-06-03 03:52:34.765627 [DEBUG] mod_freetdm.c:451 FreeTDM/1:13/7001
CHANNEL EXECUTE
2011-06-03 03:52:34.765627 [DEBUG] switch_core_state_machine.c:157
FreeTDM/1:13/7001 Standard EXECUTE
EXECUTE FreeTDM/1:13/7001 set(call_timeout=20)
2011-06-03 03:52:34.765627 [DEBUG] mod_dptools.c:1059 FreeTDM/1:13/7001 SET
[call_timeout]=[20]
EXECUTE FreeTDM/1:13/7001 info()
2011-06-03 03:52:34.765627 [INFO] mod_dptools.c:1202 CHANNEL_DATA:
Channel-State: [CS_EXECUTE]
Channel-Call-State: [RINGING]
Channel-State-Number: [4]
Channel-Name: [FreeTDM/1:13/7001]
Unique-ID: [8949f10a-ad91-4705-95a7-eb42dfcfe40f]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Channel-Call-UUID: [8949f10a-ad91-4705-95a7-eb42dfcfe40f]
Answer-State: [ringing]
Channel-Read-Codec-Name: [PCMA]
Channel-Read-Codec-Rate: [8000]
Channel-Read-Codec-Bit-Rate: [64000]
Channel-Write-Codec-Name: [PCMA]
Channel-Write-Codec-Rate: [8000]
Channel-Write-Codec-Bit-Rate: [64000]
Caller-Direction: [inbound]
Caller-Username: [FreeTDM]
Caller-Dialplan: [XML]
Caller-Caller-ID-Number: [8097328707]
Caller-ANI: [8097328707]
Caller-Destination-Number: [7001]
Caller-Unique-ID: [8949f10a-ad91-4705-95a7-eb42dfcfe40f]
Caller-Source: [mod_freetdm]
Caller-Context: [novanet]
Caller-RDNIS: [7001]
Caller-Channel-Name: [FreeTDM/1:13/7001]
Caller-Profile-Index: [2]
Caller-Profile-Created-Time: [1307073154764686]
Caller-Channel-Created-Time: [1307073154758018]
Caller-Channel-Answered-Time: [0]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
variable_direction: [inbound]
variable_uuid: [8949f10a-ad91-4705-95a7-eb42dfcfe40f]
variable_read_codec: [PCMA]
variable_read_rate: [8000]
variable_write_codec: [PCMA]
variable_write_rate: [8000]
variable_channel_name: [FreeTDM/1:13/7001]
variable_freetdm_span_name: [wp1]
variable_freetdm_span_number: [1]
variable_freetdm_chan_number: [13]
variable_freetdm_bearer_capability: [0]
variable_freetdm_bearer_layer1: [3]
variable_freetdm_screening_ind: [network-provided]
variable_freetdm_presentation_ind: [presentation-allowed]
variable_dialed_extension: [7001]
variable_transfer_ringback: [local_stream://moh]
variable_#: [true]
variable_export_vars: [#,*,#]
variable_bind_meta_key: [#]
variable_hangup_after_bridge: [true]
variable_continue_on_fail: [true]
variable_called_party_callgroup: [1]
variable_dialed_user: [7001]
variable_dialed_domain: [192.168.2.190]
variable_call_uuid: [8949f10a-ad91-4705-95a7-eb42dfcfe40f]
variable_originate_disposition: [USER_NOT_REGISTERED]
variable_max_forwards: [70]
variable_call_timeout: [20]
variable_current_application: [info]


EXECUTE FreeTDM/1:13/7001
set(continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED)
2011-06-03 03:52:34.766572 [DEBUG] mod_dptools.c:1059 FreeTDM/1:13/7001 SET
[continue_on_fail]=[NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED]
EXECUTE FreeTDM/1:13/7001 set(effective_caller_id_number=91)
2011-06-03 03:52:34.766572 [DEBUG] mod_dptools.c:1059 FreeTDM/1:13/7001 SET
[effective_caller_id_number]=[91]
EXECUTE FreeTDM/1:13/7001 ring_ready()
2011-06-03 03:52:34.766572 [DEBUG] mod_freetdm.c:861 [s1c13][1:13]
Indicating RINGING in state PROCEED
2011-06-03 03:52:34.766572 [DEBUG] mod_freetdm.c:861 [s1c13][1:13] Changed
state from PROCEED to RINGING
2011-06-03 03:52:34.766572 [DEBUG] ftdm_state.c:508 [s1c13][1:13] Executing
state processor for RINGING
2011-06-03 03:52:34.766572 [DEBUG] ftmod_sangoma_isdn.c:614 [s1c13][1:13]
Completed state change from PROCEED to RINGING in 0ms
2011-06-03 03:52:34.766572 [DEBUG] ftmod_sangoma_isdn.c:625 [s1c13][1:13]
processing state change to RINGING














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



On Thu, Jun 2, 2011 at 12:18 PM, Sam <u2nsam at gmail.com> wrote:

> Hello,
>
>
> How do i manipulate callerid number ... i have the below statement
>
> <action application="set"
> data="effective_caller_id_number=66${outbound_caller_id_number}"/>
>
> here i want to prefix caller id with 66 so the effective caller id becomes
> 66 + original callerid .
>
> using above statement the caller id passed was just 66 to the far end.
>
> below is the dialplan i used.
>
> <extension name="7001">
>             <condition field="destination_number" expression="^(7001)"
> continue="false">
>             <action application="set" data="call_timeout=20"/>
>             <action application="info" />
>             <action application="set"
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED"/>
>
>             <action application="set"
> data="effective_caller_id_number=66${outbound_caller_id_number}"/>
>             <action application="ring_ready" />
>             <action application="bridge"
> data="{sip_cid_type=rpid}sofia/gateway/outbound/1646xxxxxxx"/>
>
>         </extension>
>
>
>
>
> ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
>
> This is from header i get .
>
> From: "" <sip:66 at 203.153.53.138>
>
>
> -------------------------------------------------------------------------------------------------------------------
>
>
>
> Dialplan: FreeTDM/1:29/7001 Action info()
> Dialplan: FreeTDM/1:29/7001 Action
> set(continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED)
> Dialplan: FreeTDM/1:29/7001 Action
> set(effective_caller_id_number=66${outbound_caller_id_number})
> Dialplan: FreeTDM/1:29/7001 Action ring_ready()
> Dialplan: FreeTDM/1:29/7001 Action
> bridge({sip_cid_type=rpid}sofia/gateway/outbound/1646xxxxxxx)
> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:119
> (FreeTDM/1:29/7001) State Change CS_ROUTING -> CS_EXECUTE
> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_session.c:1116 Send signal
> FreeTDM/1:29/7001 [BREAK]
> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:359
> (FreeTDM/1:29/7001) State ROUTING going to sleep
> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:320
> (FreeTDM/1:29/7001) Running State Change CS_EXECUTE
> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:366
> (FreeTDM/1:29/7001) State EXECUTE
> 2011-06-02 06:43:48.111347 [DEBUG] mod_freetdm.c:451 FreeTDM/1:29/7001
> CHANNEL EXECUTE
> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:157
> FreeTDM/1:29/7001 Standard EXECUTE
> EXECUTE FreeTDM/1:29/7001 set(call_timeout=20)
> 2011-06-02 06:43:48.111347 [DEBUG] mod_dptools.c:1059 FreeTDM/1:29/7001 SET
> [call_timeout]=[20]
> EXECUTE FreeTDM/1:29/7001 info()
> 2011-06-02 06:43:48.111347 [INFO] mod_dptools.c:1202 CHANNEL_DATA:
> Channel-State: [CS_EXECUTE]
> Channel-Call-State: [RINGING]
> Channel-State-Number: [4]
> Channel-Name: [FreeTDM/1:29/7001]
> Unique-ID: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
> Call-Direction: [inbound]
> Presence-Call-Direction: [inbound]
> Channel-Call-UUID: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
> Answer-State: [ringing]
> Channel-Read-Codec-Name: [PCMA]
> Channel-Read-Codec-Rate: [8000]
> Channel-Read-Codec-Bit-Rate: [64000]
> Channel-Write-Codec-Name: [PCMA]
> Channel-Write-Codec-Rate: [8000]
> Channel-Write-Codec-Bit-Rate: [64000]
> Caller-Direction: [inbound]
> Caller-Username: [FreeTDM]
> Caller-Dialplan: [XML]
> Caller-Caller-ID-Number: [8097328707]
> Caller-ANI: [8097328707]
> Caller-Destination-Number: [7001]
> Caller-Unique-ID: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
> Caller-Source: [mod_freetdm]
> Caller-RDNIS: [7001]
> Caller-Channel-Name: [FreeTDM/1:29/7001]
> Caller-Profile-Index: [2]
> Caller-Profile-Created-Time: [1306997028110372]
> Caller-Channel-Created-Time: [1306997028103773]
> Caller-Channel-Answered-Time: [0]
> Caller-Channel-Progress-Time: [0]
> Caller-Channel-Progress-Media-Time: [0]
> Caller-Channel-Hangup-Time: [0]
> Caller-Channel-Transfer-Time: [0]
> Caller-Screen-Bit: [true]
> Caller-Privacy-Hide-Name: [false]
> Caller-Privacy-Hide-Number: [false]
> variable_direction: [inbound]
> variable_uuid: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
> variable_read_codec: [PCMA]
> variable_read_rate: [8000]
> variable_write_codec: [PCMA]
> variable_write_rate: [8000]
> variable_channel_name: [FreeTDM/1:29/7001]
> variable_freetdm_span_name: [wp1]
> variable_freetdm_span_number: [1]
> variable_freetdm_chan_number: [29]
> variable_freetdm_bearer_capability: [0]
> variable_freetdm_bearer_layer1: [3]
> variable_freetdm_screening_ind: [network-provided]
> variable_freetdm_presentation_ind: [presentation-allowed]
> variable_dialed_extension: [7001]
> variable_transfer_ringback: [local_stream://moh]
> variable_#: [true]
> variable_export_vars: [#,*,#]
> variable_bind_meta_key: [#]
> variable_hangup_after_bridge: [true]
> variable_continue_on_fail: [true]
> variable_called_party_callgroup: [1]
> variable_dialed_user: [7001]
> variable_dialed_domain: [192.168.2.190]
> variable_call_uuid: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
> variable_originate_disposition: [USER_NOT_REGISTERED]
> variable_max_forwards: [70]
> variable_call_timeout: [20]
> variable_current_application: [info]
>
>
>
> Regards
> Sam
>
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From jan.berger at video24.no  Fri Jun  3 09:21:18 2011
From: jan.berger at video24.no (Jan Berger)
Date: Fri, 3 Jun 2011 07:21:18 +0200
Subject: [Freeswitch-users] mrcp
In-Reply-To: <BANLkTim7qr0qbsxcjadrshjEfka_kZ+SYA@mail.gmail.com>
References: <A358176020C045358C04C08944501FAA@dell9400>
	<BANLkTi=Q6oMRUNq9GSVjMORwxS0eWV2=9w@mail.gmail.com>
	<D663E566A50142F69915A5AFB17257F7@dell9400>
	<BANLkTim7qr0qbsxcjadrshjEfka_kZ+SYA@mail.gmail.com>
Message-ID: <3B804BE828BD4B6FBC55F6FE6EBE3507@dell9400>

Thanks - i did not see that option.

 

Is it possible to do this through ESL as well?

 

Jan

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Christopher Rienzo
Sent: 3. juni 2011 04:28
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mrcp

 

To send SRGS directly, use the "inline:" prefix.   For TTS, you can send
SSML if your server supports it.

This is the APP to control speech recognition:

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_detect_speech

I need to update the wiki... it got out of sync last year when luke-jr added
support for multiple grammars per session.  

Chris



On Thu, Jun 2, 2011 at 8:07 PM, Jan Berger <jan.berger at video24.no> wrote:

Thanks,

 

It "kind of" does the job.It's nice to be able to specify URL or file as
grammar, but I need to pass SRGS or SSML directly.

 

If you attempt to send SRGS it screws up in mod_unimrcp and attempt to load
a file from what I can see - it seems to lack a test to see if it gets
passed srgs or ssml from the app directly. Need to test tomorrow.

 

Jan

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven
Ayre
Sent: 2. juni 2011 13:44
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mrcp

 

http://wiki.freeswitch.org/wiki/Mod_unimrcp

:)



On 2 June 2011 11:25, Jan Berger <jan.berger at video24.no> wrote:

Hi,

 

I would like to communicate with MRCP to control TTS and ASR from a C module
- how can I do that easiest?

 

Jan

 


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

 


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

 

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From u2nsam at gmail.com  Fri Jun  3 10:53:37 2011
From: u2nsam at gmail.com (Sam)
Date: Fri, 3 Jun 2011 12:23:37 +0530
Subject: [Freeswitch-users] domain
Message-ID: <BANLkTimO7+Sg=8cJjdfRr-1wEKgbbZDrcA@mail.gmail.com>

Hello,

I have a user registered by domain (1001 at xyz.com) and when the invite comes
to my FS server it should recognize the domain and route
the calls accordingly to the group <domain name="xyz.com">  ...   </domain>
.
The call would be entering via the acl to public.xml. Here i want to route
the call according to domain names, how will i do that.

Any suggestions...


Regards
Sam
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From steveayre at gmail.com  Fri Jun  3 11:27:12 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 3 Jun 2011 08:27:12 +0100
Subject: [Freeswitch-users] mrcp
In-Reply-To: <3B804BE828BD4B6FBC55F6FE6EBE3507@dell9400>
References: <A358176020C045358C04C08944501FAA@dell9400>
	<BANLkTi=Q6oMRUNq9GSVjMORwxS0eWV2=9w@mail.gmail.com>
	<D663E566A50142F69915A5AFB17257F7@dell9400>
	<BANLkTim7qr0qbsxcjadrshjEfka_kZ+SYA@mail.gmail.com>
	<3B804BE828BD4B6FBC55F6FE6EBE3507@dell9400>
Message-ID: <BANLkTin2QdTJp7jKW77JzhtwN+1R3P9uNw@mail.gmail.com>

You can execute an app on a channel through ESL using the uuid_broadcast
API.
http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast

-Steve


On 3 June 2011 06:21, Jan Berger <jan.berger at video24.no> wrote:

>  Thanks ? i did not see that option.
>
>
>
> Is it possible to do this through ESL as well?
>
>
>
> Jan
>
>
>  ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Christopher
> Rienzo
> *Sent:* 3. juni 2011 04:28
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] mrcp
>
>
>
> To send SRGS directly, use the "inline:" prefix.   For TTS, you can send
> SSML if your server supports it.
>
> This is the APP to control speech recognition:
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_detect_speech
>
> I need to update the wiki... it got out of sync last year when luke-jr
> added support for multiple grammars per session.
>
> Chris
>
>  On Thu, Jun 2, 2011 at 8:07 PM, Jan Berger <jan.berger at video24.no> wrote:
>
> Thanks,
>
>
>
> It ?kind of? does the job.It?s nice to be able to specify URL or file as
> grammar, but I need to pass SRGS or SSML directly.
>
>
>
> If you attempt to send SRGS it screws up in mod_unimrcp and attempt to
> load a file from what I can see ? it seems to lack a test to see if it gets
> passed srgs or ssml from the app directly. Need to test tomorrow.
>
>
>
> Jan
>  ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* 2. juni 2011 13:44
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] mrcp
>
>
>
> http://wiki.freeswitch.org/wiki/Mod_unimrcp
>
> :)
>
>  On 2 June 2011 11:25, Jan Berger <jan.berger at video24.no> wrote:
>
> Hi,
>
>
>
> I would like to communicate with MRCP to control TTS and ASR from a C
> module - how can I do that easiest?
>
>
>
> Jan
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From boris at tagnet.ru  Fri Jun  3 12:14:31 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Fri, 03 Jun 2011 14:14:31 +0600
Subject: [Freeswitch-users] manipulate CID
In-Reply-To: <BANLkTi=mDgriNWGn=7cfXXAKzwOR6EZpqA@mail.gmail.com>
References: <BANLkTi=ONknfmsocwr+4sQTK54RYUSCTcQ@mail.gmail.com>
	<BANLkTi=mDgriNWGn=7cfXXAKzwOR6EZpqA@mail.gmail.com>
Message-ID: <4DE897E7.1050705@tagnet.ru>

Hello!

     There is no variable ${Caller-Caller-ID-Number}, but 
${caller_id_number}. Please look at 
http://wiki.freeswitch.org/wiki/Channel_Variables

> I still cannot get desired caller id through prefixing,
>
> i get from header as :-  From: "" <sip:91 at xxx.xxx.xxx.xxx>
>
>
> <extension name="novanet_7001">
> <condition field="destination_number" expression="^(7001)" 
> continue="false">
> <action application="set" data="call_timeout=20"/>
> <action application="info" />
> <action application="set" 
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED"/> 
>
> <action application="set" 
> data="effective_caller_id_number=91${Caller-Caller-ID-Number}"/>
> <action application="ring_ready" />
> <action application="bridge" 
> data="{sip_cid_type=rpid}sofia/gateway/novanet/1646xxxxxxxx"/>
>
> </condition>
> </extension>
>
> Logs:
> -----------------------
>
> Dialplan: FreeTDM/1:13/7001 Action 
> set(continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED)
> Dialplan: FreeTDM/1:13/7001 Action 
> set(effective_caller_id_number=91${Caller-Caller-ID-Number})
> Dialplan: FreeTDM/1:13/7001 Action ring_ready()
> Dialplan: FreeTDM/1:13/7001 Action 
> bridge({sip_cid_type=rpid}sofia/gateway/novanet/1646xxxxxxx)
> 2011-06-03 03:52:34.765627 [DEBUG] switch_core_state_machine.c:119 
> (FreeTDM/1:13/7001) State Change CS_ROUTING -> CS_EXECUTE
> 2011-06-03 03:52:34.765627 [DEBUG] switch_core_session.c:1116 Send 
> signal FreeTDM/1:13/7001 [BREAK]
> 2011-06-03 03:52:34.765627 [DEBUG] switch_core_state_machine.c:359 
> (FreeTDM/1:13/7001) State ROUTING going to sleep
> 2011-06-03 03:52:34.765627 [DEBUG] switch_core_state_machine.c:320 
> (FreeTDM/1:13/7001) Running State Change CS_EXECUTE
> 2011-06-03 03:52:34.765627 [DEBUG] switch_core_state_machine.c:366 
> (FreeTDM/1:13/7001) State EXECUTE
> 2011-06-03 03:52:34.765627 [DEBUG] mod_freetdm.c:451 FreeTDM/1:13/7001 
> CHANNEL EXECUTE
> 2011-06-03 03:52:34.765627 [DEBUG] switch_core_state_machine.c:157 
> FreeTDM/1:13/7001 Standard EXECUTE
> EXECUTE FreeTDM/1:13/7001 set(call_timeout=20)
> 2011-06-03 03:52:34.765627 [DEBUG] mod_dptools.c:1059 
> FreeTDM/1:13/7001 SET [call_timeout]=[20]
> EXECUTE FreeTDM/1:13/7001 info()
> 2011-06-03 03:52:34.765627 [INFO] mod_dptools.c:1202 CHANNEL_DATA:
> Channel-State: [CS_EXECUTE]
> Channel-Call-State: [RINGING]
> Channel-State-Number: [4]
> Channel-Name: [FreeTDM/1:13/7001]
> Unique-ID: [8949f10a-ad91-4705-95a7-eb42dfcfe40f]
> Call-Direction: [inbound]
> Presence-Call-Direction: [inbound]
> Channel-Call-UUID: [8949f10a-ad91-4705-95a7-eb42dfcfe40f]
> Answer-State: [ringing]
> Channel-Read-Codec-Name: [PCMA]
> Channel-Read-Codec-Rate: [8000]
> Channel-Read-Codec-Bit-Rate: [64000]
> Channel-Write-Codec-Name: [PCMA]
> Channel-Write-Codec-Rate: [8000]
> Channel-Write-Codec-Bit-Rate: [64000]
> Caller-Direction: [inbound]
> Caller-Username: [FreeTDM]
> Caller-Dialplan: [XML]
> Caller-Caller-ID-Number: [8097328707]
> Caller-ANI: [8097328707]
> Caller-Destination-Number: [7001]
> Caller-Unique-ID: [8949f10a-ad91-4705-95a7-eb42dfcfe40f]
> Caller-Source: [mod_freetdm]
> Caller-Context: [novanet]
> Caller-RDNIS: [7001]
> Caller-Channel-Name: [FreeTDM/1:13/7001]
> Caller-Profile-Index: [2]
> Caller-Profile-Created-Time: [1307073154764686]
> Caller-Channel-Created-Time: [1307073154758018]
> Caller-Channel-Answered-Time: [0]
> Caller-Channel-Progress-Time: [0]
> Caller-Channel-Progress-Media-Time: [0]
> Caller-Channel-Hangup-Time: [0]
> Caller-Channel-Transfer-Time: [0]
> Caller-Screen-Bit: [true]
> Caller-Privacy-Hide-Name: [false]
> Caller-Privacy-Hide-Number: [false]
> variable_direction: [inbound]
> variable_uuid: [8949f10a-ad91-4705-95a7-eb42dfcfe40f]
> variable_read_codec: [PCMA]
> variable_read_rate: [8000]
> variable_write_codec: [PCMA]
> variable_write_rate: [8000]
> variable_channel_name: [FreeTDM/1:13/7001]
> variable_freetdm_span_name: [wp1]
> variable_freetdm_span_number: [1]
> variable_freetdm_chan_number: [13]
> variable_freetdm_bearer_capability: [0]
> variable_freetdm_bearer_layer1: [3]
> variable_freetdm_screening_ind: [network-provided]
> variable_freetdm_presentation_ind: [presentation-allowed]
> variable_dialed_extension: [7001]
> variable_transfer_ringback: [local_stream://moh]
> variable_#: [true]
> variable_export_vars: [#,*,#]
> variable_bind_meta_key: [#]
> variable_hangup_after_bridge: [true]
> variable_continue_on_fail: [true]
> variable_called_party_callgroup: [1]
> variable_dialed_user: [7001]
> variable_dialed_domain: [192.168.2.190]
> variable_call_uuid: [8949f10a-ad91-4705-95a7-eb42dfcfe40f]
> variable_originate_disposition: [USER_NOT_REGISTERED]
> variable_max_forwards: [70]
> variable_call_timeout: [20]
> variable_current_application: [info]
>
>
> EXECUTE FreeTDM/1:13/7001 
> set(continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED)
> 2011-06-03 03:52:34.766572 [DEBUG] mod_dptools.c:1059 
> FreeTDM/1:13/7001 SET 
> [continue_on_fail]=[NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED]
> EXECUTE FreeTDM/1:13/7001 set(effective_caller_id_number=91)
> 2011-06-03 03:52:34.766572 [DEBUG] mod_dptools.c:1059 
> FreeTDM/1:13/7001 SET [effective_caller_id_number]=[91]
> EXECUTE FreeTDM/1:13/7001 ring_ready()
> 2011-06-03 03:52:34.766572 [DEBUG] mod_freetdm.c:861 [s1c13][1:13] 
> Indicating RINGING in state PROCEED
> 2011-06-03 03:52:34.766572 [DEBUG] mod_freetdm.c:861 [s1c13][1:13] 
> Changed state from PROCEED to RINGING
> 2011-06-03 03:52:34.766572 [DEBUG] ftdm_state.c:508 [s1c13][1:13] 
> Executing state processor for RINGING
> 2011-06-03 03:52:34.766572 [DEBUG] ftmod_sangoma_isdn.c:614 
> [s1c13][1:13] Completed state change from PROCEED to RINGING in 0ms
> 2011-06-03 03:52:34.766572 [DEBUG] ftmod_sangoma_isdn.c:625 
> [s1c13][1:13] processing state change to RINGING
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> -----------------------------------------------------------
>
>
>
> On Thu, Jun 2, 2011 at 12:18 PM, Sam <u2nsam at gmail.com 
> <mailto:u2nsam at gmail.com>> wrote:
>
>     Hello,
>
>
>     How do i manipulate callerid number ... i have the below statement
>
>     <action application="set"
>     data="effective_caller_id_number=66${outbound_caller_id_number}"/>
>
>     here i want to prefix caller id with 66 so the effective caller id
>     becomes 66 + original callerid .
>
>     using above statement the caller id passed was just 66 to the far end.
>
>     below is the dialplan i used.
>
>     <extension name="7001">
>     <condition field="destination_number" expression="^(7001)"
>     continue="false">
>     <action application="set" data="call_timeout=20"/>
>     <action application="info" />
>     <action application="set"
>     data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED"/>
>
>     <action application="set"
>     data="effective_caller_id_number=66${outbound_caller_id_number}"/>
>     <action application="ring_ready" />
>     <action application="bridge"
>     data="{sip_cid_type=rpid}sofia/gateway/outbound/1646xxxxxxx"/>
>
>     </extension>
>
>
>
>     ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
>
>     This is from header i get .
>
>     From: "" <sip:66 at 203.153.53.138 <mailto:sip%3A66 at 203.153.53.138>>
>
>     -------------------------------------------------------------------------------------------------------------------
>
>
>
>     Dialplan: FreeTDM/1:29/7001 Action info()
>     Dialplan: FreeTDM/1:29/7001 Action
>     set(continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED)
>     Dialplan: FreeTDM/1:29/7001 Action
>     set(effective_caller_id_number=66${outbound_caller_id_number})
>     Dialplan: FreeTDM/1:29/7001 Action ring_ready()
>     Dialplan: FreeTDM/1:29/7001 Action
>     bridge({sip_cid_type=rpid}sofia/gateway/outbound/1646xxxxxxx)
>     2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:119
>     (FreeTDM/1:29/7001) State Change CS_ROUTING -> CS_EXECUTE
>     2011-06-02 06:43:48.111347 [DEBUG] switch_core_session.c:1116 Send
>     signal FreeTDM/1:29/7001 [BREAK]
>     2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:359
>     (FreeTDM/1:29/7001) State ROUTING going to sleep
>     2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:320
>     (FreeTDM/1:29/7001) Running State Change CS_EXECUTE
>     2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:366
>     (FreeTDM/1:29/7001) State EXECUTE
>     2011-06-02 06:43:48.111347 [DEBUG] mod_freetdm.c:451
>     FreeTDM/1:29/7001 CHANNEL EXECUTE
>     2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:157
>     FreeTDM/1:29/7001 Standard EXECUTE
>     EXECUTE FreeTDM/1:29/7001 set(call_timeout=20)
>     2011-06-02 06:43:48.111347 [DEBUG] mod_dptools.c:1059
>     FreeTDM/1:29/7001 SET [call_timeout]=[20]
>     EXECUTE FreeTDM/1:29/7001 info()
>     2011-06-02 06:43:48.111347 [INFO] mod_dptools.c:1202 CHANNEL_DATA:
>     Channel-State: [CS_EXECUTE]
>     Channel-Call-State: [RINGING]
>     Channel-State-Number: [4]
>     Channel-Name: [FreeTDM/1:29/7001]
>     Unique-ID: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
>     Call-Direction: [inbound]
>     Presence-Call-Direction: [inbound]
>     Channel-Call-UUID: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
>     Answer-State: [ringing]
>     Channel-Read-Codec-Name: [PCMA]
>     Channel-Read-Codec-Rate: [8000]
>     Channel-Read-Codec-Bit-Rate: [64000]
>     Channel-Write-Codec-Name: [PCMA]
>     Channel-Write-Codec-Rate: [8000]
>     Channel-Write-Codec-Bit-Rate: [64000]
>     Caller-Direction: [inbound]
>     Caller-Username: [FreeTDM]
>     Caller-Dialplan: [XML]
>     Caller-Caller-ID-Number: [8097328707]
>     Caller-ANI: [8097328707]
>     Caller-Destination-Number: [7001]
>     Caller-Unique-ID: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
>     Caller-Source: [mod_freetdm]
>     Caller-RDNIS: [7001]
>     Caller-Channel-Name: [FreeTDM/1:29/7001]
>     Caller-Profile-Index: [2]
>     Caller-Profile-Created-Time: [1306997028110372]
>     Caller-Channel-Created-Time: [1306997028103773]
>     Caller-Channel-Answered-Time: [0]
>     Caller-Channel-Progress-Time: [0]
>     Caller-Channel-Progress-Media-Time: [0]
>     Caller-Channel-Hangup-Time: [0]
>     Caller-Channel-Transfer-Time: [0]
>     Caller-Screen-Bit: [true]
>     Caller-Privacy-Hide-Name: [false]
>     Caller-Privacy-Hide-Number: [false]
>     variable_direction: [inbound]
>     variable_uuid: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
>     variable_read_codec: [PCMA]
>     variable_read_rate: [8000]
>     variable_write_codec: [PCMA]
>     variable_write_rate: [8000]
>     variable_channel_name: [FreeTDM/1:29/7001]
>     variable_freetdm_span_name: [wp1]
>     variable_freetdm_span_number: [1]
>     variable_freetdm_chan_number: [29]
>     variable_freetdm_bearer_capability: [0]
>     variable_freetdm_bearer_layer1: [3]
>     variable_freetdm_screening_ind: [network-provided]
>     variable_freetdm_presentation_ind: [presentation-allowed]
>     variable_dialed_extension: [7001]
>     variable_transfer_ringback: [local_stream://moh]
>     variable_#: [true]
>     variable_export_vars: [#,*,#]
>     variable_bind_meta_key: [#]
>     variable_hangup_after_bridge: [true]
>     variable_continue_on_fail: [true]
>     variable_called_party_callgroup: [1]
>     variable_dialed_user: [7001]
>     variable_dialed_domain: [192.168.2.190]
>     variable_call_uuid: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
>     variable_originate_disposition: [USER_NOT_REGISTERED]
>     variable_max_forwards: [70]
>     variable_call_timeout: [20]
>     variable_current_application: [info]
>
>
>
>     Regards
>     Sam
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Regards,
Boris


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From boris at tagnet.ru  Fri Jun  3 12:21:05 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Fri, 03 Jun 2011 14:21:05 +0600
Subject: [Freeswitch-users] domain
In-Reply-To: <BANLkTimO7+Sg=8cJjdfRr-1wEKgbbZDrcA@mail.gmail.com>
References: <BANLkTimO7+Sg=8cJjdfRr-1wEKgbbZDrcA@mail.gmail.com>
Message-ID: <4DE89971.9010205@tagnet.ru>

Hello!

     What do You mean? You may (for example) set the domain variable for 
a user and test it inside condition.

> Hello,
>
> I have a user registered by domain (1001 at xyz.com 
> <mailto:1001 at xyz.com>) and when the invite comes to my FS server it 
> should recognize the domain and route
> the calls accordingly to the group <domain name="xyz.com 
> <http://xyz.com>">  ... </domain> .
> The call would be entering via the acl to public.xml. Here i want to 
> route the call according to domain names, how will i do that.
>
> Any suggestions...
>
>
> Regards
> Sam
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Regards,
Boris


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From u2nsam at gmail.com  Fri Jun  3 13:03:33 2011
From: u2nsam at gmail.com (Sam)
Date: Fri, 3 Jun 2011 14:33:33 +0530
Subject: [Freeswitch-users] domain
In-Reply-To: <4DE89971.9010205@tagnet.ru>
References: <BANLkTimO7+Sg=8cJjdfRr-1wEKgbbZDrcA@mail.gmail.com>
	<4DE89971.9010205@tagnet.ru>
Message-ID: <BANLkTimOZ-yxbNXFiOr519FoR=9euOZLcg@mail.gmail.com>

The call would be coming from outside to FS and not within , so the call
would be routed from public to different domains ,

and after that following the doc
http://wiki.freeswitch.org/wiki/Multi-tenant

Here how can we recognized from the header on which domain is it ?  like if
we get an from header as 1001 at xyz.com
how can we recognized the domain and route the call accordingly ... when an
outside system sends a call via acl .

regards
Sam

On Fri, Jun 3, 2011 at 1:51 PM, Boris Kovalenko <boris at tagnet.ru> wrote:

>  Hello!
>
>     What do You mean? You may (for example) set the domain variable for a
> user and test it inside condition.
>
> Hello,
>
> I have a user registered by domain (1001 at xyz.com) and when the invite
> comes to my FS server it should recognize the domain and route
> the calls accordingly to the group <domain name="xyz.com">  ...
> </domain> .
> The call would be entering via the acl to public.xml. Here i want to route
> the call according to domain names, how will i do that.
>
> Any suggestions...
>
>
> Regards
> Sam
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> Regards,
> Boris
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Fri Jun  3 13:04:44 2011
From: u2nsam at gmail.com (Sam)
Date: Fri, 3 Jun 2011 14:34:44 +0530
Subject: [Freeswitch-users] manipulate CID
In-Reply-To: <4DE897E7.1050705@tagnet.ru>
References: <BANLkTi=ONknfmsocwr+4sQTK54RYUSCTcQ@mail.gmail.com>
	<BANLkTi=mDgriNWGn=7cfXXAKzwOR6EZpqA@mail.gmail.com>
	<4DE897E7.1050705@tagnet.ru>
Message-ID: <BANLkTi=LzOCcRWT998VhQ8dZQ6VU6ka4Dw@mail.gmail.com>

Thanks it works

On Fri, Jun 3, 2011 at 1:44 PM, Boris Kovalenko <boris at tagnet.ru> wrote:

>  Hello!
>
>     There is no variable ${Caller-Caller-ID-Number}, but
> ${caller_id_number}. Please look at
> http://wiki.freeswitch.org/wiki/Channel_Variables
>
> I still cannot get desired caller id through prefixing,
>
> i get from header as :-  From: "" <sip:91 at xxx.xxx.xxx.xxx>
>
>
> <extension name="novanet_7001">
>             <condition field="destination_number" expression="^(7001)"
> continue="false">
>             <action application="set" data="call_timeout=20"/>
>             <action application="info" />
>             <action application="set"
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED"/>
>
>             <action application="set"
> data="effective_caller_id_number=91${Caller-Caller-ID-Number}"/>
>             <action application="ring_ready" />
>             <action application="bridge"
> data="{sip_cid_type=rpid}sofia/gateway/novanet/1646xxxxxxxx"/>
>
> </condition>
>         </extension>
>
> Logs:
> -----------------------
>
> Dialplan: FreeTDM/1:13/7001 Action
> set(continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED)
> Dialplan: FreeTDM/1:13/7001 Action
> set(effective_caller_id_number=91${Caller-Caller-ID-Number})
> Dialplan: FreeTDM/1:13/7001 Action ring_ready()
> Dialplan: FreeTDM/1:13/7001 Action
> bridge({sip_cid_type=rpid}sofia/gateway/novanet/1646xxxxxxx)
> 2011-06-03 03:52:34.765627 [DEBUG] switch_core_state_machine.c:119
> (FreeTDM/1:13/7001) State Change CS_ROUTING -> CS_EXECUTE
> 2011-06-03 03:52:34.765627 [DEBUG] switch_core_session.c:1116 Send signal
> FreeTDM/1:13/7001 [BREAK]
> 2011-06-03 03:52:34.765627 [DEBUG] switch_core_state_machine.c:359
> (FreeTDM/1:13/7001) State ROUTING going to sleep
> 2011-06-03 03:52:34.765627 [DEBUG] switch_core_state_machine.c:320
> (FreeTDM/1:13/7001) Running State Change CS_EXECUTE
> 2011-06-03 03:52:34.765627 [DEBUG] switch_core_state_machine.c:366
> (FreeTDM/1:13/7001) State EXECUTE
> 2011-06-03 03:52:34.765627 [DEBUG] mod_freetdm.c:451 FreeTDM/1:13/7001
> CHANNEL EXECUTE
> 2011-06-03 03:52:34.765627 [DEBUG] switch_core_state_machine.c:157
> FreeTDM/1:13/7001 Standard EXECUTE
> EXECUTE FreeTDM/1:13/7001 set(call_timeout=20)
> 2011-06-03 03:52:34.765627 [DEBUG] mod_dptools.c:1059 FreeTDM/1:13/7001 SET
> [call_timeout]=[20]
> EXECUTE FreeTDM/1:13/7001 info()
> 2011-06-03 03:52:34.765627 [INFO] mod_dptools.c:1202 CHANNEL_DATA:
> Channel-State: [CS_EXECUTE]
> Channel-Call-State: [RINGING]
> Channel-State-Number: [4]
> Channel-Name: [FreeTDM/1:13/7001]
> Unique-ID: [8949f10a-ad91-4705-95a7-eb42dfcfe40f]
> Call-Direction: [inbound]
> Presence-Call-Direction: [inbound]
> Channel-Call-UUID: [8949f10a-ad91-4705-95a7-eb42dfcfe40f]
> Answer-State: [ringing]
> Channel-Read-Codec-Name: [PCMA]
> Channel-Read-Codec-Rate: [8000]
> Channel-Read-Codec-Bit-Rate: [64000]
> Channel-Write-Codec-Name: [PCMA]
> Channel-Write-Codec-Rate: [8000]
> Channel-Write-Codec-Bit-Rate: [64000]
> Caller-Direction: [inbound]
> Caller-Username: [FreeTDM]
> Caller-Dialplan: [XML]
> Caller-Caller-ID-Number: [8097328707]
> Caller-ANI: [8097328707]
> Caller-Destination-Number: [7001]
> Caller-Unique-ID: [8949f10a-ad91-4705-95a7-eb42dfcfe40f]
> Caller-Source: [mod_freetdm]
> Caller-Context: [novanet]
> Caller-RDNIS: [7001]
> Caller-Channel-Name: [FreeTDM/1:13/7001]
> Caller-Profile-Index: [2]
> Caller-Profile-Created-Time: [1307073154764686]
> Caller-Channel-Created-Time: [1307073154758018]
> Caller-Channel-Answered-Time: [0]
> Caller-Channel-Progress-Time: [0]
> Caller-Channel-Progress-Media-Time: [0]
> Caller-Channel-Hangup-Time: [0]
> Caller-Channel-Transfer-Time: [0]
> Caller-Screen-Bit: [true]
> Caller-Privacy-Hide-Name: [false]
> Caller-Privacy-Hide-Number: [false]
> variable_direction: [inbound]
> variable_uuid: [8949f10a-ad91-4705-95a7-eb42dfcfe40f]
> variable_read_codec: [PCMA]
> variable_read_rate: [8000]
> variable_write_codec: [PCMA]
> variable_write_rate: [8000]
> variable_channel_name: [FreeTDM/1:13/7001]
> variable_freetdm_span_name: [wp1]
> variable_freetdm_span_number: [1]
> variable_freetdm_chan_number: [13]
> variable_freetdm_bearer_capability: [0]
> variable_freetdm_bearer_layer1: [3]
> variable_freetdm_screening_ind: [network-provided]
> variable_freetdm_presentation_ind: [presentation-allowed]
> variable_dialed_extension: [7001]
> variable_transfer_ringback: [local_stream://moh]
> variable_#: [true]
> variable_export_vars: [#,*,#]
> variable_bind_meta_key: [#]
> variable_hangup_after_bridge: [true]
> variable_continue_on_fail: [true]
> variable_called_party_callgroup: [1]
> variable_dialed_user: [7001]
> variable_dialed_domain: [192.168.2.190]
> variable_call_uuid: [8949f10a-ad91-4705-95a7-eb42dfcfe40f]
> variable_originate_disposition: [USER_NOT_REGISTERED]
> variable_max_forwards: [70]
> variable_call_timeout: [20]
> variable_current_application: [info]
>
>
> EXECUTE FreeTDM/1:13/7001
> set(continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED)
> 2011-06-03 03:52:34.766572 [DEBUG] mod_dptools.c:1059 FreeTDM/1:13/7001 SET
> [continue_on_fail]=[NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED]
> EXECUTE FreeTDM/1:13/7001 set(effective_caller_id_number=91)
> 2011-06-03 03:52:34.766572 [DEBUG] mod_dptools.c:1059 FreeTDM/1:13/7001 SET
> [effective_caller_id_number]=[91]
> EXECUTE FreeTDM/1:13/7001 ring_ready()
> 2011-06-03 03:52:34.766572 [DEBUG] mod_freetdm.c:861 [s1c13][1:13]
> Indicating RINGING in state PROCEED
> 2011-06-03 03:52:34.766572 [DEBUG] mod_freetdm.c:861 [s1c13][1:13] Changed
> state from PROCEED to RINGING
> 2011-06-03 03:52:34.766572 [DEBUG] ftdm_state.c:508 [s1c13][1:13] Executing
> state processor for RINGING
> 2011-06-03 03:52:34.766572 [DEBUG] ftmod_sangoma_isdn.c:614 [s1c13][1:13]
> Completed state change from PROCEED to RINGING in 0ms
> 2011-06-03 03:52:34.766572 [DEBUG] ftmod_sangoma_isdn.c:625 [s1c13][1:13]
> processing state change to RINGING
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> -----------------------------------------------------------
>
>
>
> On Thu, Jun 2, 2011 at 12:18 PM, Sam <u2nsam at gmail.com> wrote:
>
>> Hello,
>>
>>
>> How do i manipulate callerid number ... i have the below statement
>>
>> <action application="set"
>> data="effective_caller_id_number=66${outbound_caller_id_number}"/>
>>
>> here i want to prefix caller id with 66 so the effective caller id becomes
>> 66 + original callerid .
>>
>> using above statement the caller id passed was just 66 to the far end.
>>
>> below is the dialplan i used.
>>
>> <extension name="7001">
>>             <condition field="destination_number" expression="^(7001)"
>> continue="false">
>>             <action application="set" data="call_timeout=20"/>
>>             <action application="info" />
>>             <action application="set"
>> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED"/>
>>
>>             <action application="set"
>> data="effective_caller_id_number=66${outbound_caller_id_number}"/>
>>             <action application="ring_ready" />
>>             <action application="bridge"
>> data="{sip_cid_type=rpid}sofia/gateway/outbound/1646xxxxxxx"/>
>>
>>         </extension>
>>
>>
>>
>>
>> ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
>>
>> This is from header i get .
>>
>> From: "" <sip:66 at 203.153.53.138>
>>
>>
>> -------------------------------------------------------------------------------------------------------------------
>>
>>
>>
>> Dialplan: FreeTDM/1:29/7001 Action info()
>> Dialplan: FreeTDM/1:29/7001 Action
>> set(continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED)
>> Dialplan: FreeTDM/1:29/7001 Action
>> set(effective_caller_id_number=66${outbound_caller_id_number})
>> Dialplan: FreeTDM/1:29/7001 Action ring_ready()
>> Dialplan: FreeTDM/1:29/7001 Action
>> bridge({sip_cid_type=rpid}sofia/gateway/outbound/1646xxxxxxx)
>> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:119
>> (FreeTDM/1:29/7001) State Change CS_ROUTING -> CS_EXECUTE
>> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_session.c:1116 Send signal
>> FreeTDM/1:29/7001 [BREAK]
>> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:359
>> (FreeTDM/1:29/7001) State ROUTING going to sleep
>> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:320
>> (FreeTDM/1:29/7001) Running State Change CS_EXECUTE
>> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:366
>> (FreeTDM/1:29/7001) State EXECUTE
>> 2011-06-02 06:43:48.111347 [DEBUG] mod_freetdm.c:451 FreeTDM/1:29/7001
>> CHANNEL EXECUTE
>> 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:157
>> FreeTDM/1:29/7001 Standard EXECUTE
>> EXECUTE FreeTDM/1:29/7001 set(call_timeout=20)
>> 2011-06-02 06:43:48.111347 [DEBUG] mod_dptools.c:1059 FreeTDM/1:29/7001
>> SET [call_timeout]=[20]
>> EXECUTE FreeTDM/1:29/7001 info()
>> 2011-06-02 06:43:48.111347 [INFO] mod_dptools.c:1202 CHANNEL_DATA:
>> Channel-State: [CS_EXECUTE]
>> Channel-Call-State: [RINGING]
>> Channel-State-Number: [4]
>> Channel-Name: [FreeTDM/1:29/7001]
>> Unique-ID: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
>> Call-Direction: [inbound]
>> Presence-Call-Direction: [inbound]
>> Channel-Call-UUID: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
>> Answer-State: [ringing]
>> Channel-Read-Codec-Name: [PCMA]
>> Channel-Read-Codec-Rate: [8000]
>> Channel-Read-Codec-Bit-Rate: [64000]
>> Channel-Write-Codec-Name: [PCMA]
>> Channel-Write-Codec-Rate: [8000]
>> Channel-Write-Codec-Bit-Rate: [64000]
>> Caller-Direction: [inbound]
>> Caller-Username: [FreeTDM]
>> Caller-Dialplan: [XML]
>> Caller-Caller-ID-Number: [8097328707]
>> Caller-ANI: [8097328707]
>> Caller-Destination-Number: [7001]
>> Caller-Unique-ID: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
>> Caller-Source: [mod_freetdm]
>> Caller-RDNIS: [7001]
>> Caller-Channel-Name: [FreeTDM/1:29/7001]
>> Caller-Profile-Index: [2]
>> Caller-Profile-Created-Time: [1306997028110372]
>> Caller-Channel-Created-Time: [1306997028103773]
>> Caller-Channel-Answered-Time: [0]
>> Caller-Channel-Progress-Time: [0]
>> Caller-Channel-Progress-Media-Time: [0]
>> Caller-Channel-Hangup-Time: [0]
>> Caller-Channel-Transfer-Time: [0]
>> Caller-Screen-Bit: [true]
>> Caller-Privacy-Hide-Name: [false]
>> Caller-Privacy-Hide-Number: [false]
>> variable_direction: [inbound]
>> variable_uuid: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
>> variable_read_codec: [PCMA]
>> variable_read_rate: [8000]
>> variable_write_codec: [PCMA]
>> variable_write_rate: [8000]
>> variable_channel_name: [FreeTDM/1:29/7001]
>> variable_freetdm_span_name: [wp1]
>> variable_freetdm_span_number: [1]
>> variable_freetdm_chan_number: [29]
>> variable_freetdm_bearer_capability: [0]
>> variable_freetdm_bearer_layer1: [3]
>> variable_freetdm_screening_ind: [network-provided]
>> variable_freetdm_presentation_ind: [presentation-allowed]
>> variable_dialed_extension: [7001]
>> variable_transfer_ringback: [local_stream://moh]
>> variable_#: [true]
>> variable_export_vars: [#,*,#]
>> variable_bind_meta_key: [#]
>> variable_hangup_after_bridge: [true]
>> variable_continue_on_fail: [true]
>> variable_called_party_callgroup: [1]
>> variable_dialed_user: [7001]
>> variable_dialed_domain: [192.168.2.190]
>> variable_call_uuid: [e5b47553-3b9c-4692-beaa-513a3a3356c2]
>> variable_originate_disposition: [USER_NOT_REGISTERED]
>> variable_max_forwards: [70]
>> variable_call_timeout: [20]
>> variable_current_application: [info]
>>
>>
>>
>> Regards
>> Sam
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> Regards,
> Boris
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sharad at coraltele.com  Fri Jun  3 13:16:58 2011
From: sharad at coraltele.com (sharad)
Date: Fri, 3 Jun 2011 14:46:58 +0530
Subject: [Freeswitch-users] core.db location
References: <mailman.10041.1307091918.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <3396F43C6C34416BAD6E25D2702995B2@sharad>

Dear All

Is there any way to host only core.db in some other predefined directory ?

Let all other db remain in the folder db.

Plz advise.

Regards
Ankur




----- Original Message ----- 
From: <freeswitch-users-request at lists.freeswitch.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Friday, June 03, 2011 2:35 PM
Subject: FreeSWITCH-users Digest, Vol 60, Issue 21


> Send FreeSWITCH-users mailing list submissions to
> freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
> freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
> freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>


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


> Today's Topics:
>
>   1. Re: domain (Boris Kovalenko)
>   2. Re: domain (Sam)
>   3. Re: manipulate CID (Sam)
>


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


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



From boris at tagnet.ru  Fri Jun  3 13:16:48 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Fri, 03 Jun 2011 15:16:48 +0600
Subject: [Freeswitch-users] domain
In-Reply-To: <BANLkTimOZ-yxbNXFiOr519FoR=9euOZLcg@mail.gmail.com>
References: <BANLkTimO7+Sg=8cJjdfRr-1wEKgbbZDrcA@mail.gmail.com>	<4DE89971.9010205@tagnet.ru>
	<BANLkTimOZ-yxbNXFiOr519FoR=9euOZLcg@mail.gmail.com>
Message-ID: <4DE8A680.1080102@tagnet.ru>

Hello!

     There is a variable named ${sip_from_user}. You may do something like

<action application="set" 
data="from_domain=${regex(${sip_from_user}|^.*@(.*)$|%1)}" inline="true"/>
<action application="transfer" data="${from_domain} XML domain_context"/>

> The call would be coming from outside to FS and not within , so the 
> call would be routed from public to different domains ,
>
> and after that following the doc 
> http://wiki.freeswitch.org/wiki/Multi-tenant
>
> Here how can we recognized from the header on which domain is it ?  
> like if we get an from header as 1001 at xyz.com <mailto:1001 at xyz.com>
> how can we recognized the domain and route the call accordingly ... 
> when an outside system sends a call via acl .
>
> regards
> Sam
>
> On Fri, Jun 3, 2011 at 1:51 PM, Boris Kovalenko <boris at tagnet.ru 
> <mailto:boris at tagnet.ru>> wrote:
>
>     Hello!
>
>         What do You mean? You may (for example) set the domain
>     variable for a user and test it inside condition.
>
>>     Hello,
>>
>>     I have a user registered by domain (1001 at xyz.com
>>     <mailto:1001 at xyz.com>) and when the invite comes to my FS server
>>     it should recognize the domain and route
>>     the calls accordingly to the group <domain name="xyz.com
>>     <http://xyz.com>">  ... </domain> .
>>     The call would be entering via the acl to public.xml. Here i want
>>     to route the call according to domain names, how will i do that.
>>
>>     Any suggestions...
>>
>>
>>     Regards
>>     Sam
>>
>>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>     -- 
>     Regards,
>     Boris
>
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Regards,
Boris


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From fdelawarde at wirelessmundi.com  Fri Jun  3 13:20:39 2011
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Fri, 03 Jun 2011 11:20:39 +0200
Subject: [Freeswitch-users] skype reverse engineered
Message-ID: <1307092839.6781.599.camel@luna.madrid.commsmundi.com>

Maybe it could be used to improve mod_skypopen??

http://skype-open-source.blogspot.com/2011/06/skype-protocol-reverse-engineered.html


Fran?ois.



From thomas at chaschperli.ch  Fri Jun  3 13:32:48 2011
From: thomas at chaschperli.ch (Thomas Mueller)
Date: Fri, 03 Jun 2011 11:32:48 +0200
Subject: [Freeswitch-users] core.db location
In-Reply-To: <3396F43C6C34416BAD6E25D2702995B2@sharad>
References: <mailman.10041.1307091918.24555.freeswitch-users@lists.freeswitch.org>
	<3396F43C6C34416BAD6E25D2702995B2@sharad>
Message-ID: <4DE8AA40.40608@chaschperli.ch>

On 03.06.2011 11:16, sharad wrote:
> Dear All
>
> Is there any way to host only core.db in some other predefined directory ?
>
> Let all other db remain in the folder db.
>

on Linux you could symlink the core.db file to where you want it.

- Thomas


From thomas at chaschperli.ch  Fri Jun  3 14:15:01 2011
From: thomas at chaschperli.ch (Thomas Mueller)
Date: Fri, 03 Jun 2011 12:15:01 +0200
Subject: [Freeswitch-users] core.db location
In-Reply-To: <655CA25849B34C998B7A1F14B2514DF3@sharad>
References: <655CA25849B34C998B7A1F14B2514DF3@sharad>
Message-ID: <4DE8B425.7000002@chaschperli.ch>


> Just tried it. Symlink is created but the issue is when freeswitch
> starts, it removes the symlink. So how to handle this ?

cannot confirm this:

* stopped freeswitch
* moved core.db to core.db.symlinked
* ln -s core.db.symlinked core.db
* start freeswitch
* file-list:
root       root        17 Jun  3 12:11 core.db -> core.db.symlinked
freeswitch daemon  144384 Jun  3 12:11 core.db.symlinked

core.db still symlinked to core.db.symlinked

- Thomas



From steveayre at gmail.com  Fri Jun  3 14:19:10 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 3 Jun 2011 11:19:10 +0100
Subject: [Freeswitch-users] skype reverse engineered
In-Reply-To: <1307092839.6781.599.camel@luna.madrid.commsmundi.com>
References: <1307092839.6781.599.camel@luna.madrid.commsmundi.com>
Message-ID: <BANLkTikob7yF3ZP1yjB8QX9eTFC0EnfjzQ@mail.gmail.com>

Since it'd allow native support while skypopen requires the official client,
it'd be a big enough rewrite to probably be more of a separate module.

It remains to see what the M$ reaction will be to that project. It sounds
fairly easy for them to break support for it too. Even if they don't go the
legal route, it's an older version of the protocol and they could just
choose to block those older versions and force users to upgrade.

-Steve



On 3 June 2011 10:20, Fran?ois Delawarde <fdelawarde at wirelessmundi.com>wrote:

> Maybe it could be used to improve mod_skypopen??
>
>
> http://skype-open-source.blogspot.com/2011/06/skype-protocol-reverse-engineered.html
>
>
> Fran?ois.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Fri Jun  3 14:25:22 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 3 Jun 2011 11:25:22 +0100
Subject: [Freeswitch-users] skype reverse engineered
In-Reply-To: <BANLkTikob7yF3ZP1yjB8QX9eTFC0EnfjzQ@mail.gmail.com>
References: <1307092839.6781.599.camel@luna.madrid.commsmundi.com>
	<BANLkTikob7yF3ZP1yjB8QX9eTFC0EnfjzQ@mail.gmail.com>
Message-ID: <BANLkTiktSoB_qUWjWvyRrq3NFD3CUT6UWQ@mail.gmail.com>

However interestingly it appears to be 1.x, 3.x and 4.x have reverse
engineered. 5.x is the latest windows version.

The latest Linux version (used by mod_skypopen) is 2.2.0.25. Not sure why
that's not reverse engineered but if they were to block anything pre-5.x
it'd break both skyopen and skype-open-source.

On the more optimistic note I have seen a good argument that there are
enough hardware Skype phones with Skype that they can't easily block older
versions without breaking a lot of devices.

-Steve


On 3 June 2011 11:19, Steven Ayre <steveayre at gmail.com> wrote:

> Since it'd allow native support while skypopen requires the official
> client, it'd be a big enough rewrite to probably be more of a separate
> module.
>
> It remains to see what the M$ reaction will be to that project. It sounds
> fairly easy for them to break support for it too. Even if they don't go the
> legal route, it's an older version of the protocol and they could just
> choose to block those older versions and force users to upgrade.
>
> -Steve
>
>
>
>
> On 3 June 2011 10:20, Fran?ois Delawarde <fdelawarde at wirelessmundi.com>wrote:
>
>> Maybe it could be used to improve mod_skypopen??
>>
>>
>> http://skype-open-source.blogspot.com/2011/06/skype-protocol-reverse-engineered.html
>>
>>
>> Fran?ois.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From sharad at coraltele.com  Fri Jun  3 14:59:16 2011
From: sharad at coraltele.com (sharad)
Date: Fri, 3 Jun 2011 16:29:16 +0530
Subject: [Freeswitch-users] core.db location
References: <655CA25849B34C998B7A1F14B2514DF3@sharad>
	<4DE8B425.7000002@chaschperli.ch>
Message-ID: <CAF01D3B367F436EBBB07665AD952FD2@sharad>

Sorry Mr. Thomas

I made a mistake in testing. Its working.

Thanks a lot for your kind advise.

Best Regards
Sharad

----- Original Message ----- 
From: "Thomas Mueller" <thomas at chaschperli.ch>
To: "sharad" <sharad at coraltele.com>
Cc: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Friday, June 03, 2011 3:45 PM
Subject: Re: [Freeswitch-users] core.db location


> 
>> Just tried it. Symlink is created but the issue is when freeswitch
>> starts, it removes the symlink. So how to handle this ?
> 
> cannot confirm this:
> 
> * stopped freeswitch
> * moved core.db to core.db.symlinked
> * ln -s core.db.symlinked core.db
> * start freeswitch
> * file-list:
> root       root        17 Jun  3 12:11 core.db -> core.db.symlinked
> freeswitch daemon  144384 Jun  3 12:11 core.db.symlinked
> 
> core.db still symlinked to core.db.symlinked
> 
> - Thomas
> 
> 
> 
>


From freeswitch at peely.com  Fri Jun  3 16:03:48 2011
From: freeswitch at peely.com (peely)
Date: Fri, 3 Jun 2011 05:03:48 -0700 (PDT)
Subject: [Freeswitch-users] ptime mismatch confusion
In-Reply-To: <BANLkTinUxgFMJjbuMwV=_MbHh09hMk9wvQ@mail.gmail.com>
References: <BANLkTim-3A4s9G3eLvHTEssg8=xf=-ysCA@mail.gmail.com>
	<BANLkTimd3fyz3S0ATTTSq19gM+Nthdtdzg@mail.gmail.com>
	<BANLkTin3HpV-7UnCUiiHLoYHSQBFksx4hQ@mail.gmail.com>
	<BANLkTinUxgFMJjbuMwV=_MbHh09hMk9wvQ@mail.gmail.com>
Message-ID: <1307102628006-6434987.post@n2.nabble.com>

You should really be able to set a channel variable in the dialplan just
before you handle the call with <action application="set"
data="absolute_codec_string=PCMA at 20i"/> which will set the a-leg codecs and
ptimes, then in the bridge add absolute_codec_string=${global_codec_prefs}
to set the codecs to use on the b-leg.




--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ptime-mismatch-confusion-tp6432622p6434987.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble at slickdeals.endjunk.com  Fri Jun  3 16:28:52 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 3 Jun 2011 05:28:52 -0700 (PDT)
Subject: [Freeswitch-users] skype reverse engineered
In-Reply-To: <BANLkTiktSoB_qUWjWvyRrq3NFD3CUT6UWQ@mail.gmail.com>
References: <1307092839.6781.599.camel@luna.madrid.commsmundi.com>
	<BANLkTikob7yF3ZP1yjB8QX9eTFC0EnfjzQ@mail.gmail.com>
	<BANLkTiktSoB_qUWjWvyRrq3NFD3CUT6UWQ@mail.gmail.com>
Message-ID: <1307104132721-6435062.post@n2.nabble.com>


Steven Ayre wrote:
> On the more optimistic note I have seen a good argument that there are
> enough hardware Skype phones with Skype that they can't easily block older
> versions without breaking a lot of devices.
If Skype decides to block older version of software, then all it needs is to
furnish Skype Phone hardware manufacturers with a new code to come up with a
new firmware to support new Skype version. Then, hackers will start to
reversed engineer on the new version. And, the cat/mouse chase begins.

OTOH, it looks like the source was released to 
https://github.com/skypeopensource/skypeopensource/downloads GitHUB  on
6/2/11. let's see how fast it will become a Linux compilable
package/plugin/module. I won't be surprised to see a working
mod_skype_re(versed engineered) soon if Anthony and/or FS developers will
decide to port the source into an FS module.

-----
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From leonardo.bidinoto at voicetechnology.com.br  Fri Jun  3 16:31:44 2011
From: leonardo.bidinoto at voicetechnology.com.br (Leonardo P. Bidinoto)
Date: Fri, 3 Jun 2011 09:31:44 -0300
Subject: [Freeswitch-users] Hanged up callers doubt
In-Reply-To: <2C18EE80-BB65-4A93-9A44-627B6154DD5F@ipeva.fr>
References: <BANLkTi=qf4T3Wb9_2TmJdt4Wwif5qipsmg@mail.gmail.com>
	<BANLkTik+DYhPT1ss8-OqpQsvUAA30A5L=Q@mail.gmail.com>
	<BANLkTinxTu=c_9usFNv0sOOWdQ3-DnAYXQ@mail.gmail.com>
	<BANLkTi=g3J1nZTKArouGSJMDHVSE1L0idA@mail.gmail.com>
	<BANLkTi=WN97QMJcpwyEGQNvRxBW1c=UsuQ@mail.gmail.com>
	<BANLkTiky4M2ZgTMS228-DJ6oaEsmkpG19g@mail.gmail.com>
	<BANLkTimzsNxVyFpLf6x8X4E7P58FYwdHng@mail.gmail.com>
	<BANLkTinYynZsX6=zg-yyeP7hEQ7-q5J9+Q@mail.gmail.com>
	<BANLkTim=ueSmFA_ug2MbbLFMBh0362_2Sg@mail.gmail.com>
	<BANLkTikV_-=Zkv8zOQVcWyg5Gc06UtRxjw@mail.gmail.com>
	<2C18EE80-BB65-4A93-9A44-627B6154DD5F@ipeva.fr>
Message-ID: <BANLkTimYO9kV2VFcraTCjn4uLOtAu44R3Q@mail.gmail.com>

Hi David,

Yes, indeed that should work, BUT it doesn't if an user inside is stucked(i
get a "false" from "uuid_exists").
With "conference kick", all users inside will be dropped, but the stucked
ones will remain, and with that the conference isnt destroyed.



2011/6/2 David Ponzone <david.ponzone at ipeva.fr>

> Sure, that one is easy:
>
> from fs_cli:
>
> conference <confname> kick all
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 02/06/2011 ? 21:33, Leonardo P. Bidinoto a ?crit :
>
> Hi,
>
> About this issue, could be a way to destroy the conference from FS, in a
> attempt to remove the stucked channels from the "show channels " and
> "conference list"?
>
> 2011/5/17 Leonardo P. Bidinoto <leonardo.bidinoto at voicetechnology.com.br>
>
>> Sure. Im sending a pcap file made by tcpdump and one that i made by ngrep.
>> In both files, it was registering whats happening when i stuck the channel
>> by hanging up while using a ESL connection inside a conference(app socket
>> 8085 sync full). I did a "conference kick" command in this channel while its
>> was waiting to close the ESL connection.
>>
>>
>>
>> 2011/5/16 Michael Collins <msc at freeswitch.org>
>>
>>> Can you tcpdump or otherwise capture the traffic on port 8085? I am
>>> curious what is happening with that.
>>> -MC
>>>
>>>
>>> On Mon, May 16, 2011 at 12:12 PM, Leonardo P. Bidinoto <
>>> leonardo.bidinoto at voicetechnology.com.br> wrote:
>>>
>>>> hehe, ok michael.
>>>>
>>>> here is the pastebin link:
>>>> http://pastebin.freeswitch.org/16303
>>>>
>>>>
>>>> 2011/5/13 Michael Collins <msc at freeswitch.org>
>>>>
>>>>> Pastebin this info and select "FreeSWITCH Log" as the syntax
>>>>> highlighting. I need the colorized output to read logs. (I'm getting older
>>>>> and it's hard for me to ready black and white in an email.)
>>>>>
>>>>> -MC
>>>>>
>>>>>
>>>>> On Fri, May 13, 2011 at 7:32 AM, Leonardo P. Bidinoto <
>>>>> leonardo.bidinoto at voicetechnology.com.br> wrote:
>>>>>
>>>>>> Hi Michael,
>>>>>>
>>>>>> Just succeeded to reproduce the problem.
>>>>>>
>>>>>> The condition is: when a channel inside a conference is using a ESL
>>>>>> connection(lets call it "A") through socket application and another ESL
>>>>>> connection(lets call it "B") executes a command with this channel, the "B"
>>>>>> ESL connection will wait the "A" ESL connection close to execute its
>>>>>> command.
>>>>>> If the channel hangs up before the "A" ESL connection is closed, then
>>>>>> "B" ESL command will never be executed and the stucked channel will still be
>>>>>> there, into sofia and the conference too.
>>>>>> To verify that, just do "show channels" and "conference list". with
>>>>>> "uuid_exists" command, return "false".
>>>>>>
>>>>>> Here are the actions done by the channel before get stucked:
>>>>>>
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.187321
>>>>>> [NOTICE] switch_channel.c:816 New Channel sofia/external/
>>>>>> 1000123402 at 192.168.0.154 [16e09413-9cb0-4011-a635-f91933a35c0f]
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>> [DEBUG] sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154entering state [received][100]
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>> [DEBUG] sofia.c:4772 Remote SDP:
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>> [DEBUG] sofia_glue.c:4656 Audio Codec Compare
>>>>>> [BV32:107:16000:20:0]/[PCMU:0:8000:20:64000]
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>> [DEBUG] sofia_glue.c:4656 Audio Codec Compare
>>>>>> [BV32:107:16000:20:0]/[PCMA:8:8000:20:64000]
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>> [DEBUG] sofia_glue.c:4656 Audio Codec Compare
>>>>>> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>> [DEBUG] sofia_glue.c:2788 Set Codec sofia/external/
>>>>>> 1000123402 at 192.168.0.154 PCMU/8000 20 ms 160 samples 64000 bits
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>> [DEBUG] sofia_glue.c:4770 Set 2833 dtmf send/recv payload to 101
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>> [DEBUG] sofia.c:4943 (sofia/external/1000123402 at 192.168.0.154) State
>>>>>> Change CS_NEW -> CS_INIT
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>> [DEBUG] switch_core_session.c:1114 Send signal sofia/external/
>>>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>> [DEBUG] switch_core_state_machine.c:325 (sofia/external/
>>>>>> 1000123402 at 192.168.0.154) Running State Change CS_INIT
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>> [DEBUG] switch_core_state_machine.c:361 (sofia/external/
>>>>>> 1000123402 at 192.168.0.154) State INIT
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>> [DEBUG] mod_sofia.c:84 sofia/external/1000123402 at 192.168.0.154 SOFIA
>>>>>> INIT
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>> [DEBUG] mod_sofia.c:124 (sofia/external/1000123402 at 192.168.0.154)
>>>>>> State Change CS_INIT -> CS_ROUTING
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>> [DEBUG] switch_core_session.c:1114 Send signal sofia/external/
>>>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>> [DEBUG] switch_core_state_machine.c:361 (sofia/external/
>>>>>> 1000123402 at 192.168.0.154) State INIT going to sleep
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>> [DEBUG] switch_core_state_machine.c:325 (sofia/external/
>>>>>> 1000123402 at 192.168.0.154) Running State Change CS_ROUTING
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>> [DEBUG] switch_channel.c:1665 (sofia/external/
>>>>>> 1000123402 at 192.168.0.154) Callstate Change DOWN -> RINGING
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>> [DEBUG] switch_core_state_machine.c:364 (sofia/external/
>>>>>> 1000123402 at 192.168.0.154) State ROUTING
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>> [DEBUG] mod_sofia.c:147 sofia/external/1000123402 at 192.168.0.154 SOFIA
>>>>>> ROUTING
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>> [DEBUG] switch_core_state_machine.c:77 sofia/external/
>>>>>> 1000123402 at 192.168.0.154 Standard ROUTING
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [INFO]
>>>>>> mod_dialplan_xml.c:331 Processing Phone1 <1000123402>->1234567890 in context
>>>>>> public
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>>> 1000123402 at 192.168.0.154 parsing [public->unloop] continue=false
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>>> 1000123402 at 192.168.0.154 Regex (PASS) [unloop] ${unroll_loops}(true)
>>>>>> =~ /^true$/ break=on-false
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>>> 1000123402 at 192.168.0.154 Regex (FAIL) [unloop] ${sip_looped_call}()
>>>>>> =~ /^true$/ break=on-false
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>>> 1000123402 at 192.168.0.154 parsing [public->public_extensions]
>>>>>> continue=false
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>>> 1000123402 at 192.168.0.154 Regex (PASS) [public_extensions]
>>>>>> destination_number(1234567890) =~ /^(\d*)$/ break=on-false
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>>> 1000123402 at 192.168.0.154 Action transfer(1234567890 XML default)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328
>>>>>> [DEBUG] switch_core_state_machine.c:119 (sofia/external/
>>>>>> 1000123402 at 192.168.0.154) State Change CS_ROUTING -> CS_EXECUTE
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328
>>>>>> [DEBUG] switch_core_session.c:1114 Send signal sofia/external/
>>>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328
>>>>>> [DEBUG] switch_core_state_machine.c:364 (sofia/external/
>>>>>> 1000123402 at 192.168.0.154) State ROUTING going to sleep
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328
>>>>>> [DEBUG] switch_core_state_machine.c:325 (sofia/external/
>>>>>> 1000123402 at 192.168.0.154) Running State Change CS_EXECUTE
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328
>>>>>> [DEBUG] switch_core_state_machine.c:371 (sofia/external/
>>>>>> 1000123402 at 192.168.0.154) State EXECUTE
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328
>>>>>> [DEBUG] mod_sofia.c:240 sofia/external/1000123402 at 192.168.0.154 SOFIA
>>>>>> EXECUTE
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327
>>>>>> [DEBUG] switch_core_state_machine.c:157 sofia/external/
>>>>>> 1000123402 at 192.168.0.154 Standard EXECUTE
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>> 1000123402 at 192.168.0.154 transfer(1234567890 XML default)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327
>>>>>> [DEBUG] switch_ivr.c:1597 (sofia/external/1000123402 at 192.168.0.154)
>>>>>> State Change CS_EXECUTE -> CS_ROUTING
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327
>>>>>> [DEBUG] switch_core_session.c:1114 Send signal sofia/external/
>>>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327
>>>>>> [DEBUG] switch_core_session.c:707 Send signal sofia/external/
>>>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327
>>>>>> [NOTICE] switch_ivr.c:1603 Transfer sofia/external/
>>>>>> 1000123402 at 192.168.0.154 to XML[1234567890 at default]
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327
>>>>>> [DEBUG] switch_core_state_machine.c:371 (sofia/external/
>>>>>> 1000123402 at 192.168.0.154) State EXECUTE going to sleep
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327
>>>>>> [DEBUG] switch_core_state_machine.c:325 (sofia/external/
>>>>>> 1000123402 at 192.168.0.154) Running State Change CS_ROUTING
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327
>>>>>> [DEBUG] switch_core_state_machine.c:364 (sofia/external/
>>>>>> 1000123402 at 192.168.0.154) State ROUTING
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327
>>>>>> [DEBUG] mod_sofia.c:147 sofia/external/1000123402 at 192.168.0.154 SOFIA
>>>>>> ROUTING
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327
>>>>>> [DEBUG] switch_core_state_machine.c:77 sofia/external/
>>>>>> 1000123402 at 192.168.0.154 Standard ROUTING
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [INFO]
>>>>>> mod_dialplan_xml.c:331 Processing Phone1 <1000123402>->1234567890 in context
>>>>>> default
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>>> 1000123402 at 192.168.0.154 parsing [default->flex] continue=false
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>>> 1000123402 at 192.168.0.154 Regex (PASS) [flex]
>>>>>> destination_number(1234567890) =~ /^(\d+)$/ break=on-false
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>>> 1000123402 at 192.168.0.154 Action log(INFO VOICE received
>>>>>> dest=1234567890)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>>> 1000123402 at 192.168.0.154 Action set(playback_terminators=#)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>>> 1000123402 at 192.168.0.154 Action log(INFO Let's do some ivrd, shall
>>>>>> we?)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>>> 1000123402 at 192.168.0.154 Action
>>>>>> set(ivr_path=/usr/local/ipconf/fs/daemon/ivr.rb)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>>> 1000123402 at 192.168.0.154 Action socket(localhost:8084 full)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325
>>>>>> [DEBUG] switch_core_state_machine.c:119 (sofia/external/
>>>>>> 1000123402 at 192.168.0.154) State Change CS_ROUTING -> CS_EXECUTE
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325
>>>>>> [DEBUG] switch_core_session.c:1114 Send signal sofia/external/
>>>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325
>>>>>> [DEBUG] switch_core_state_machine.c:364 (sofia/external/
>>>>>> 1000123402 at 192.168.0.154) State ROUTING going to sleep
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325
>>>>>> [DEBUG] switch_core_state_machine.c:325 (sofia/external/
>>>>>> 1000123402 at 192.168.0.154) Running State Change CS_EXECUTE
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325
>>>>>> [DEBUG] switch_core_state_machine.c:371 (sofia/external/
>>>>>> 1000123402 at 192.168.0.154) State EXECUTE
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325
>>>>>> [DEBUG] mod_sofia.c:240 sofia/external/1000123402 at 192.168.0.154 SOFIA
>>>>>> EXECUTE
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325
>>>>>> [DEBUG] switch_core_state_machine.c:157 sofia/external/
>>>>>> 1000123402 at 192.168.0.154 Standard EXECUTE
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>> 1000123402 at 192.168.0.154 log(INFO VOICE received dest=1234567890)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [INFO]
>>>>>> mod_dptools.c:1184 VOICE received dest=1234567890
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>> 1000123402 at 192.168.0.154 set(playback_terminators=#)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331
>>>>>> [DEBUG] mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154SET [playback_terminators]=[#]
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>> 1000123402 at 192.168.0.154 log(INFO Let's do some ivrd, shall we?)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331 [INFO]
>>>>>> mod_dptools.c:1184 Let's do some ivrd, shall we?
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>> 1000123402 at 192.168.0.154set(ivr_path=/usr/local/ipconf/fs/daemon/ivr.rb)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331
>>>>>> [DEBUG] mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154SET [ivr_path]=[/usr/local/ipconf/fs/daemon/ivr.rb]
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>> 1000123402 at 192.168.0.154 socket(localhost:8084 full)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309
>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute answer()
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>> 1000123402 at 192.168.0.154 answer()
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309
>>>>>> [DEBUG] sofia_glue.c:3022 AUDIO RTP [sofia/external/
>>>>>> 1000123402 at 192.168.0.154] 192.168.0.154 port 24232 -> 192.168.0.111
>>>>>> port 4046 codec: 0 ms: 20
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309
>>>>>> [DEBUG] switch_rtp.c:1623 Starting timer [soft] 160 bytes per 20ms
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313
>>>>>> [DEBUG] sofia_glue.c:3284 Set 2833 dtmf send payload to 101
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313
>>>>>> [DEBUG] sofia_glue.c:3289 Set 2833 dtmf receive payload to 101
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313
>>>>>> [DEBUG] mod_sofia.c:681 Local SDP sofia/external/
>>>>>> 1000123402 at 192.168.0.154:
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313
>>>>>> [DEBUG] switch_core_session.c:707 Send signal sofia/external/
>>>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313
>>>>>> [DEBUG] switch_channel.c:2827 (sofia/external/
>>>>>> 1000123402 at 192.168.0.154) Callstate Change RINGING -> ACTIVE
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313
>>>>>> [NOTICE] mod_dptools.c:930 Channel [sofia/external/
>>>>>> 1000123402 at 192.168.0.154] has been answered
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.933312
>>>>>> [DEBUG] sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154entering state [completed][200]
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.040285
>>>>>> [DEBUG] sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154entering state [ready][200]
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.082294
>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute
>>>>>> playback(/usr/local/freeswitch/sounds/flex/app32/teste/WelcomeToTheConferenceSystem.wav)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>> 1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/WelcomeToTheConferenceSystem.wav)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.082294
>>>>>> [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1
>>>>>> channels 20ms
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.114277
>>>>>> [DEBUG] switch_ivr_play_say.c:1649 done playing file
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.155281
>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute read(1 1
>>>>>> /usr/local/freeswitch/sounds/flex/app32/teste/Mainmenu.wav flex_digits 5000
>>>>>> )
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>> 1000123402 at 192.168.0.154 read(1 1
>>>>>> /usr/local/freeswitch/sounds/flex/app32/teste/Mainmenu.wav flex_digits 5000
>>>>>> )
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.155281
>>>>>> [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1
>>>>>> channels 20ms
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.893670
>>>>>> [DEBUG] switch_rtp.c:3280 RTP RECV DTMF 1:1120
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.893670
>>>>>> [DEBUG] switch_ivr_play_say.c:1649 done playing file
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.897674
>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute set(flex_digits)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>> 1000123402 at 192.168.0.154 set(flex_digits)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.897674
>>>>>> [DEBUG] mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154SET [flex_digits]=[UNDEF]
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.940673
>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute read(1 1
>>>>>> /usr/local/freeswitch/sounds/flex/app32/teste/PleaseEnterYourPassCodeNow.wav
>>>>>> flex_digits 5000 )
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>> 1000123402 at 192.168.0.154 read(1 1
>>>>>> /usr/local/freeswitch/sounds/flex/app32/teste/PleaseEnterYourPassCodeNow.wav
>>>>>> flex_digits 5000 )
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.940673
>>>>>> [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1
>>>>>> channels 20ms
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.113662
>>>>>> [DEBUG] switch_rtp.c:3280 RTP RECV DTMF 8:640
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.113662
>>>>>> [DEBUG] switch_ivr_play_say.c:1649 done playing file
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.116663
>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute set(flex_digits)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>> 1000123402 at 192.168.0.154 set(flex_digits)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.116663
>>>>>> [DEBUG] mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154SET [flex_digits]=[UNDEF]
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.117663
>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute read(1 11
>>>>>> /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 5000
>>>>>> #,*)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>> 1000123402 at 192.168.0.154 read(1 11
>>>>>> /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 5000
>>>>>> #,*)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.117663
>>>>>> [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1
>>>>>> channels 20ms
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.153658
>>>>>> [DEBUG] switch_ivr_play_say.c:1649 done playing file
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.053615
>>>>>> [DEBUG] switch_rtp.c:3280 RTP RECV DTMF #:960
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.055618
>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute set(flex_digits)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>> 1000123402 at 192.168.0.154 set(flex_digits)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.055618
>>>>>> [DEBUG] mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154SET [flex_digits]=[UNDEF]
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.146613
>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute
>>>>>> playback(/usr/local/freeswitch/sounds/flex/app32/teste/InternalHelp.wav)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>> 1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/InternalHelp.wav)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.146613
>>>>>> [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1
>>>>>> channels 20ms
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603
>>>>>> [DEBUG] switch_rtp.c:3280 RTP RECV DTMF #:800
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603
>>>>>> [DEBUG] mod_dptools.c:1664 Digit #
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603
>>>>>> [DEBUG] switch_ivr_play_say.c:1649 done playing file
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.350617
>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute
>>>>>> playback(/usr/local/freeswitch/sounds/flex/app32/teste/YouAreTheOnlyPersonOnThisConference.wav)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>> 1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/YouAreTheOnlyPersonOnThisConference.wav)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.350617
>>>>>> [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1
>>>>>> channels 20ms
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.253520
>>>>>> [DEBUG] switch_ivr_play_say.c:1649 done playing file
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.322510
>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute conference(15646 at teste
>>>>>> +flags{waste})
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>> 1000123402 at 192.168.0.154 conference(15646 at teste+flags{waste})
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492
>>>>>> [DEBUG] mod_conference.c:5582 Raw Codec Activation Success L16 at 8000hz1 channel 20ms
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492
>>>>>> [DEBUG] mod_conference.c:5627 Raw Codec Activation Success L16 at 8000hz1 channel 20ms
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492
>>>>>> [DEBUG] switch_core_codec.c:116 sofia/external/
>>>>>> 1000123402 at 192.168.0.154 Push codec L16:70
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:05.280469
>>>>>> [DEBUG] switch_core_session.c:707 Send signal sofia/external/
>>>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:05.280469
>>>>>> [DEBUG] mod_conference.c:2557 Setup timer soft success interval: 20
>>>>>> samples: 160
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.573928
>>>>>> [DEBUG] switch_rtp.c:3280 RTP RECV DTMF *:960
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.593929
>>>>>> [DEBUG]  mod_conference.c:2021 Execute app: socket, localhost:8085 sync full
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.593929
>>>>>> [DEBUG] switch_core_codec.c:141 sofia/external/
>>>>>> 1000123402 at 192.168.0.154 Restore previous codec PCMU:0.
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>> 1000123402 at 192.168.0.154 socket(localhost:8085 sync full)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.097927
>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute read(1 1
>>>>>> /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 )
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>> 1000123402 at 192.168.0.154 read(1 1
>>>>>> /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 )
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.097927
>>>>>> [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1
>>>>>> channels 20ms
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.133923
>>>>>> [DEBUG] switch_ivr_play_say.c:1649 done playing file
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.373913
>>>>>> [DEBUG] switch_rtp.c:3280 RTP RECV DTMF 1:1120
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.375912
>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute set(flex_digits)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>> 1000123402 at 192.168.0.154 set(flex_digits)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.375912
>>>>>> [DEBUG] mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154SET [flex_digits]=[UNDEF]
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.490911
>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute
>>>>>> playback(/usr/local/freeswitch/sounds/flex/app32/teste/SelfMuteOn.wav)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>> 1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/SelfMuteOn.wav)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.490911
>>>>>> [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1
>>>>>> channels 20ms
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.233824
>>>>>> [DEBUG] switch_ivr_play_say.c:1649 done playing file
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.235828
>>>>>> [DEBUG] switch_cpp.cpp:988 sofia/external/1000123402 at 192.168.0.154destroy/unlink session from object
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.235828
>>>>>> [DEBUG] switch_core_codec.c:116 sofia/external/
>>>>>> 1000123402 at 192.168.0.154 Push codec L16:70
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.433668
>>>>>> [DEBUG] switch_rtp.c:3280 RTP RECV DTMF *:800
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.453667
>>>>>> [DEBUG]  mod_conference.c:2021 Execute app: socket, localhost:8085 sync full
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.453667
>>>>>> [DEBUG] switch_core_codec.c:141 sofia/external/
>>>>>> 1000123402 at 192.168.0.154 Restore previous codec PCMU:0.
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>> 1000123402 at 192.168.0.154 socket(localhost:8085 sync full)
>>>>>>
>>>>>> ====================================================================================================================================================
>>>>>> While Inside this connection, a "conference 15646 kick [member_id of
>>>>>> this channels]" command is executed by a fs_cli console and get stuck while
>>>>>> waiting response.
>>>>>>
>>>>>> ====================================================================================================================================================
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.000850
>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute read(1 1
>>>>>> /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 )
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>> 1000123402 at 192.168.0.154 read(1 1
>>>>>> /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 )
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.000850
>>>>>> [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1
>>>>>> channels 20ms
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.033846
>>>>>> [DEBUG] switch_ivr_play_say.c:1649 done playing file
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.553821
>>>>>> [DEBUG] switch_rtp.c:3280 RTP RECV DTMF 1:960
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.555825
>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute set(flex_digits)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>> 1000123402 at 192.168.0.154 set(flex_digits)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.555825
>>>>>> [DEBUG] mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154SET [flex_digits]=[UNDEF]
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244
>>>>>> [DEBUG] switch_channel.c:2560 (sofia/external/
>>>>>> 1000123402 at 192.168.0.154) Callstate Change ACTIVE -> HANGUP
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244
>>>>>> [NOTICE] sofia.c:538 Hangup sofia/external/1000123402 at 192.168.0.154[CS_EXECUTE] [NORMAL_CLEARING]
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244
>>>>>> [DEBUG] switch_channel.c:2576 Send signal sofia/external/
>>>>>> 1000123402 at 192.168.0.154 [KILL]
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244
>>>>>> [DEBUG] switch_core_session.c:1114 Send signal sofia/external/
>>>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246
>>>>>> [DEBUG] switch_core_session.c:2057 sofia/external/
>>>>>> 1000123402 at 192.168.0.154 skip receive message
>>>>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246
>>>>>> [DEBUG] switch_cpp.cpp:988 sofia/external/1000123402 at 192.168.0.154destroy/unlink session from object
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246
>>>>>> [DEBUG] switch_core_session.c:2057 sofia/external/
>>>>>> 1000123402 at 192.168.0.154 skip receive message
>>>>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246
>>>>>> [DEBUG] switch_core_codec.c:116 sofia/external/
>>>>>> 1000123402 at 192.168.0.154 Push codec L16:70
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246
>>>>>> [DEBUG] mod_conference.c:2815 Channel leaving conference, cause:
>>>>>> NORMAL_CLEARING
>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.757239
>>>>>> [DEBUG] mod_conference.c:6104 sofia/external/1000123402 at 192.168.0.154skip receive message [UNBRIDGE] (channel is hungup already)
>>>>>>
>>>>>> I hope this info helps.
>>>>>>
>>>>>> 2011/5/12 Michael Collins <msc at freeswitch.org>
>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Thu, May 12, 2011 at 12:27 PM, Leonardo P. Bidinoto <
>>>>>>> leonardo.bidinoto at voicetechnology.com.br> wrote:
>>>>>>>
>>>>>>>> Hi Michael,
>>>>>>>>
>>>>>>>> Im not using to any cdr module.
>>>>>>>
>>>>>>>
>>>>>>> I would recommend that you do several things:
>>>>>>>
>>>>>>> #1 - update to latest git
>>>>>>> #2 - rotate logs
>>>>>>> #3 - enable uuid logging (see logfile.conf.xml - param name "uuid")
>>>>>>> #4 - reproduce the symptom with a single call (if possible)
>>>>>>> #5 - pastebin the log for the uuid in question and link to it in this
>>>>>>> thread
>>>>>>>
>>>>>>> From there hopefully we'll get a clue as to what is happening.
>>>>>>> -MC
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Leonardo Pires Bidinoto
>>>>>> Voice Technology
>>>>>> www.voicetechnology.com.br
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
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>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Leonardo Pires Bidinoto
>>>> Voice Technology
>>>> www.voicetechnology.com.br
>>>>
>>>> _______________________________________________
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>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>>
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>>
>>
>> --
>> Leonardo Pires Bidinoto
>> Voice Technology
>> www.voicetechnology.com.br
>>
>
>
>
> --
> Leonardo Pires Bidinoto
> Voice Technology
> www.voicetechnology.com.br
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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-- 
Leonardo Pires Bidinoto
Voice Technology
www.voicetechnology.com.br
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From david.ponzone at ipeva.fr  Fri Jun  3 16:41:34 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 3 Jun 2011 14:41:34 +0200
Subject: [Freeswitch-users] Hanged up callers doubt
In-Reply-To: <BANLkTimYO9kV2VFcraTCjn4uLOtAu44R3Q@mail.gmail.com>
References: <BANLkTi=qf4T3Wb9_2TmJdt4Wwif5qipsmg@mail.gmail.com>
	<BANLkTik+DYhPT1ss8-OqpQsvUAA30A5L=Q@mail.gmail.com>
	<BANLkTinxTu=c_9usFNv0sOOWdQ3-DnAYXQ@mail.gmail.com>
	<BANLkTi=g3J1nZTKArouGSJMDHVSE1L0idA@mail.gmail.com>
	<BANLkTi=WN97QMJcpwyEGQNvRxBW1c=UsuQ@mail.gmail.com>
	<BANLkTiky4M2ZgTMS228-DJ6oaEsmkpG19g@mail.gmail.com>
	<BANLkTimzsNxVyFpLf6x8X4E7P58FYwdHng@mail.gmail.com>
	<BANLkTinYynZsX6=zg-yyeP7hEQ7-q5J9+Q@mail.gmail.com>
	<BANLkTim=ueSmFA_ug2MbbLFMBh0362_2Sg@mail.gmail.com>
	<BANLkTikV_-=Zkv8zOQVcWyg5Gc06UtRxjw@mail.gmail.com>
	<2C18EE80-BB65-4A93-9A44-627B6154DD5F@ipeva.fr>
	<BANLkTimYO9kV2VFcraTCjn4uLOtAu44R3Q@mail.gmail.com>
Message-ID: <793379B8-D1FC-47F4-8387-4E2789A69C72@ipeva.fr>

I guess there is no way to drop stuck calls. That's why they are stuck.
I think you should find a way to solve the root cause.

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

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

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




Le 03/06/2011 ? 14:31, Leonardo P. Bidinoto a ?crit :

> Hi David,
> 
> Yes, indeed that should work, BUT it doesn't if an user inside is stucked(i get a "false" from "uuid_exists").
> With "conference kick", all users inside will be dropped, but the stucked ones will remain, and with that the conference isnt destroyed.
> 
> 
> 
> 2011/6/2 David Ponzone <david.ponzone at ipeva.fr>
> Sure, that one is easy:
> 
> from fs_cli:
> 
> conference <confname> kick all
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 02/06/2011 ? 21:33, Leonardo P. Bidinoto a ?crit :
> 
>> Hi,
>> 
>> About this issue, could be a way to destroy the conference from FS, in a attempt to remove the stucked channels from the "show channels " and "conference list"?
>> 
>> 2011/5/17 Leonardo P. Bidinoto <leonardo.bidinoto at voicetechnology.com.br>
>> Sure. Im sending a pcap file made by tcpdump and one that i made by ngrep. In both files, it was registering whats happening when i stuck the channel by hanging up while using a ESL connection inside a conference(app socket 8085 sync full). I did a "conference kick" command in this channel while its was waiting to close the ESL connection.
>> 
>> 
>> 
>> 2011/5/16 Michael Collins <msc at freeswitch.org>
>> Can you tcpdump or otherwise capture the traffic on port 8085? I am curious what is happening with that.
>> -MC
>> 
>> 
>> On Mon, May 16, 2011 at 12:12 PM, Leonardo P. Bidinoto <leonardo.bidinoto at voicetechnology.com.br> wrote:
>> hehe, ok michael.
>> 
>> here is the pastebin link:
>> http://pastebin.freeswitch.org/16303
>> 
>> 
>> 2011/5/13 Michael Collins <msc at freeswitch.org>
>> Pastebin this info and select "FreeSWITCH Log" as the syntax highlighting. I need the colorized output to read logs. (I'm getting older and it's hard for me to ready black and white in an email.)
>> 
>> -MC
>> 
>> 
>> On Fri, May 13, 2011 at 7:32 AM, Leonardo P. Bidinoto <leonardo.bidinoto at voicetechnology.com.br> wrote:
>> Hi Michael,
>> 
>> Just succeeded to reproduce the problem.
>> 
>> The condition is: when a channel inside a conference is using a ESL connection(lets call it "A") through socket application and another ESL connection(lets call it "B") executes a command with this channel, the "B" ESL connection will wait the "A" ESL connection close to execute its command.
>> If the channel hangs up before the "A" ESL connection is closed, then "B" ESL command will never be executed and the stucked channel will still be there, into sofia and the conference too.
>> To verify that, just do "show channels" and "conference list". with "uuid_exists" command, return "false".
>> 
>> Here are the actions done by the channel before get stucked:
>> 
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.187321 [NOTICE] switch_channel.c:816 New Channel sofia/external/1000123402 at 192.168.0.154 [16e09413-9cb0-4011-a635-f91933a35c0f]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154 entering state [received][100]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] sofia.c:4772 Remote SDP:
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] sofia_glue.c:4656 Audio Codec Compare [BV32:107:16000:20:0]/[PCMU:0:8000:20:64000]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] sofia_glue.c:4656 Audio Codec Compare [BV32:107:16000:20:0]/[PCMA:8:8000:20:64000]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] sofia_glue.c:4656 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] sofia_glue.c:2788 Set Codec sofia/external/1000123402 at 192.168.0.154 PCMU/8000 20 ms 160 samples 64000 bits
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] sofia_glue.c:4770 Set 2833 dtmf send/recv payload to 101
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] sofia.c:4943 (sofia/external/1000123402 at 192.168.0.154) State Change CS_NEW -> CS_INIT
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/1000123402 at 192.168.0.154 [BREAK]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154) Running State Change CS_INIT
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] switch_core_state_machine.c:361 (sofia/external/1000123402 at 192.168.0.154) State INIT
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] mod_sofia.c:84 sofia/external/1000123402 at 192.168.0.154 SOFIA INIT
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] mod_sofia.c:124 (sofia/external/1000123402 at 192.168.0.154) State Change CS_INIT -> CS_ROUTING
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/1000123402 at 192.168.0.154 [BREAK]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] switch_core_state_machine.c:361 (sofia/external/1000123402 at 192.168.0.154) State INIT going to sleep
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154) Running State Change CS_ROUTING
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] switch_channel.c:1665 (sofia/external/1000123402 at 192.168.0.154) Callstate Change DOWN -> RINGING
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] switch_core_state_machine.c:364 (sofia/external/1000123402 at 192.168.0.154) State ROUTING
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] mod_sofia.c:147 sofia/external/1000123402 at 192.168.0.154 SOFIA ROUTING
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [DEBUG] switch_core_state_machine.c:77 sofia/external/1000123402 at 192.168.0.154 Standard ROUTING
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323 [INFO] mod_dialplan_xml.c:331 Processing Phone1 <1000123402>->1234567890 in context public
>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/1000123402 at 192.168.0.154 parsing [public->unloop] continue=false
>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/1000123402 at 192.168.0.154 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/1000123402 at 192.168.0.154 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/1000123402 at 192.168.0.154 parsing [public->public_extensions] continue=false
>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/1000123402 at 192.168.0.154 Regex (PASS) [public_extensions] destination_number(1234567890) =~ /^(\d*)$/ break=on-false
>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/1000123402 at 192.168.0.154 Action transfer(1234567890 XML default) 
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG] switch_core_state_machine.c:119 (sofia/external/1000123402 at 192.168.0.154) State Change CS_ROUTING -> CS_EXECUTE
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/1000123402 at 192.168.0.154 [BREAK]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG] switch_core_state_machine.c:364 (sofia/external/1000123402 at 192.168.0.154) State ROUTING going to sleep
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG] switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154) Running State Change CS_EXECUTE
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG] switch_core_state_machine.c:371 (sofia/external/1000123402 at 192.168.0.154) State EXECUTE
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328 [DEBUG] mod_sofia.c:240 sofia/external/1000123402 at 192.168.0.154 SOFIA EXECUTE
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG] switch_core_state_machine.c:157 sofia/external/1000123402 at 192.168.0.154 Standard EXECUTE
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 transfer(1234567890 XML default)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG] switch_ivr.c:1597 (sofia/external/1000123402 at 192.168.0.154) State Change CS_EXECUTE -> CS_ROUTING
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/1000123402 at 192.168.0.154 [BREAK]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG] switch_core_session.c:707 Send signal sofia/external/1000123402 at 192.168.0.154 [BREAK]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [NOTICE] switch_ivr.c:1603 Transfer sofia/external/1000123402 at 192.168.0.154 to XML[1234567890 at default]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG] switch_core_state_machine.c:371 (sofia/external/1000123402 at 192.168.0.154) State EXECUTE going to sleep
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG] switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154) Running State Change CS_ROUTING
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG] switch_core_state_machine.c:364 (sofia/external/1000123402 at 192.168.0.154) State ROUTING
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG] mod_sofia.c:147 sofia/external/1000123402 at 192.168.0.154 SOFIA ROUTING
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [DEBUG] switch_core_state_machine.c:77 sofia/external/1000123402 at 192.168.0.154 Standard ROUTING
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327 [INFO] mod_dialplan_xml.c:331 Processing Phone1 <1000123402>->1234567890 in context default
>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/1000123402 at 192.168.0.154 parsing [default->flex] continue=false
>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/1000123402 at 192.168.0.154 Regex (PASS) [flex] destination_number(1234567890) =~ /^(\d+)$/ break=on-false
>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/1000123402 at 192.168.0.154 Action log(INFO VOICE received dest=1234567890) 
>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/1000123402 at 192.168.0.154 Action set(playback_terminators=#) 
>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/1000123402 at 192.168.0.154 Action log(INFO Let's do some ivrd, shall we?) 
>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/1000123402 at 192.168.0.154 Action set(ivr_path=/usr/local/ipconf/fs/daemon/ivr.rb) 
>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/1000123402 at 192.168.0.154 Action socket(localhost:8084 full) 
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG] switch_core_state_machine.c:119 (sofia/external/1000123402 at 192.168.0.154) State Change CS_ROUTING -> CS_EXECUTE
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/1000123402 at 192.168.0.154 [BREAK]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG] switch_core_state_machine.c:364 (sofia/external/1000123402 at 192.168.0.154) State ROUTING going to sleep
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG] switch_core_state_machine.c:325 (sofia/external/1000123402 at 192.168.0.154) Running State Change CS_EXECUTE
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG] switch_core_state_machine.c:371 (sofia/external/1000123402 at 192.168.0.154) State EXECUTE
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG] mod_sofia.c:240 sofia/external/1000123402 at 192.168.0.154 SOFIA EXECUTE
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [DEBUG] switch_core_state_machine.c:157 sofia/external/1000123402 at 192.168.0.154 Standard EXECUTE
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 log(INFO VOICE received dest=1234567890)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325 [INFO] mod_dptools.c:1184 VOICE received dest=1234567890
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 set(playback_terminators=#)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331 [DEBUG] mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET [playback_terminators]=[#]
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 log(INFO Let's do some ivrd, shall we?)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331 [INFO] mod_dptools.c:1184 Let's do some ivrd, shall we?
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 set(ivr_path=/usr/local/ipconf/fs/daemon/ivr.rb)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331 [DEBUG] mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET [ivr_path]=[/usr/local/ipconf/fs/daemon/ivr.rb]
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 socket(localhost:8084 full)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute answer()
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 answer()
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309 [DEBUG] sofia_glue.c:3022 AUDIO RTP [sofia/external/1000123402 at 192.168.0.154] 192.168.0.154 port 24232 -> 192.168.0.111 port 4046 codec: 0 ms: 20
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309 [DEBUG] switch_rtp.c:1623 Starting timer [soft] 160 bytes per 20ms
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG] sofia_glue.c:3284 Set 2833 dtmf send payload to 101
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG] sofia_glue.c:3289 Set 2833 dtmf receive payload to 101
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG] mod_sofia.c:681 Local SDP sofia/external/1000123402 at 192.168.0.154:
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG] switch_core_session.c:707 Send signal sofia/external/1000123402 at 192.168.0.154 [BREAK]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [DEBUG] switch_channel.c:2827 (sofia/external/1000123402 at 192.168.0.154) Callstate Change RINGING -> ACTIVE
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313 [NOTICE] mod_dptools.c:930 Channel [sofia/external/1000123402 at 192.168.0.154] has been answered
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.933312 [DEBUG] sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154 entering state [completed][200]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.040285 [DEBUG] sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154 entering state [ready][200]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.082294 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute playback(/usr/local/freeswitch/sounds/flex/app32/teste/WelcomeToTheConferenceSystem.wav)
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 playback(/usr/local/freeswitch/sounds/flex/app32/teste/WelcomeToTheConferenceSystem.wav)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.082294 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.114277 [DEBUG] switch_ivr_play_say.c:1649 done playing file
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.155281 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Mainmenu.wav flex_digits 5000 )
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Mainmenu.wav flex_digits 5000 )
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.155281 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.893670 [DEBUG] switch_rtp.c:3280 RTP RECV DTMF 1:1120
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.893670 [DEBUG] switch_ivr_play_say.c:1649 done playing file
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.897674 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute set(flex_digits)
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 set(flex_digits)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.897674 [DEBUG] mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET [flex_digits]=[UNDEF]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.940673 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/PleaseEnterYourPassCodeNow.wav flex_digits 5000 )
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/PleaseEnterYourPassCodeNow.wav flex_digits 5000 )
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.940673 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.113662 [DEBUG] switch_rtp.c:3280 RTP RECV DTMF 8:640
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.113662 [DEBUG] switch_ivr_play_say.c:1649 done playing file
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.116663 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute set(flex_digits)
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 set(flex_digits)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.116663 [DEBUG] mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET [flex_digits]=[UNDEF]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.117663 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute read(1 11 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 5000 #,*)
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 read(1 11 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 5000 #,*)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.117663 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.153658 [DEBUG] switch_ivr_play_say.c:1649 done playing file
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.053615 [DEBUG] switch_rtp.c:3280 RTP RECV DTMF #:960
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.055618 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute set(flex_digits)
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 set(flex_digits)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.055618 [DEBUG] mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET [flex_digits]=[UNDEF]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.146613 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute playback(/usr/local/freeswitch/sounds/flex/app32/teste/InternalHelp.wav)
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 playback(/usr/local/freeswitch/sounds/flex/app32/teste/InternalHelp.wav)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.146613 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603 [DEBUG] switch_rtp.c:3280 RTP RECV DTMF #:800
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603 [DEBUG] mod_dptools.c:1664 Digit #
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603 [DEBUG] switch_ivr_play_say.c:1649 done playing file
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.350617 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute playback(/usr/local/freeswitch/sounds/flex/app32/teste/YouAreTheOnlyPersonOnThisConference.wav)
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 playback(/usr/local/freeswitch/sounds/flex/app32/teste/YouAreTheOnlyPersonOnThisConference.wav)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.350617 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.253520 [DEBUG] switch_ivr_play_say.c:1649 done playing file
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.322510 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute conference(15646 at teste+flags{waste})
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 conference(15646 at teste+flags{waste})
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492 [DEBUG] mod_conference.c:5582 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492 [DEBUG] mod_conference.c:5627 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492 [DEBUG] switch_core_codec.c:116 sofia/external/1000123402 at 192.168.0.154 Push codec L16:70
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:05.280469 [DEBUG] switch_core_session.c:707 Send signal sofia/external/1000123402 at 192.168.0.154 [BREAK]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:05.280469 [DEBUG] mod_conference.c:2557 Setup timer soft success interval: 20  samples: 160
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.573928 [DEBUG] switch_rtp.c:3280 RTP RECV DTMF *:960
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.593929 [DEBUG]  mod_conference.c:2021 Execute app: socket, localhost:8085 sync full
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.593929 [DEBUG] switch_core_codec.c:141 sofia/external/1000123402 at 192.168.0.154 Restore previous codec PCMU:0.
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 socket(localhost:8085 sync full)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.097927 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 )
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 )
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.097927 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.133923 [DEBUG] switch_ivr_play_say.c:1649 done playing file
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.373913 [DEBUG] switch_rtp.c:3280 RTP RECV DTMF 1:1120
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.375912 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute set(flex_digits)
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 set(flex_digits)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.375912 [DEBUG] mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET [flex_digits]=[UNDEF]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.490911 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute playback(/usr/local/freeswitch/sounds/flex/app32/teste/SelfMuteOn.wav)
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 playback(/usr/local/freeswitch/sounds/flex/app32/teste/SelfMuteOn.wav)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.490911 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.233824 [DEBUG] switch_ivr_play_say.c:1649 done playing file
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.235828 [DEBUG] switch_cpp.cpp:988 sofia/external/1000123402 at 192.168.0.154 destroy/unlink session from object
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.235828 [DEBUG] switch_core_codec.c:116 sofia/external/1000123402 at 192.168.0.154 Push codec L16:70
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.433668 [DEBUG] switch_rtp.c:3280 RTP RECV DTMF *:800
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.453667 [DEBUG]  mod_conference.c:2021 Execute app: socket, localhost:8085 sync full
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.453667 [DEBUG] switch_core_codec.c:141 sofia/external/1000123402 at 192.168.0.154 Restore previous codec PCMU:0.
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 socket(localhost:8085 sync full)
>> ====================================================================================================================================================
>> While Inside this connection, a "conference 15646 kick [member_id of this channels]" command is executed by a fs_cli console and get stuck while waiting response.
>> ====================================================================================================================================================
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.000850 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 )
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 read(1 1 /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 )
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.000850 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.033846 [DEBUG] switch_ivr_play_say.c:1649 done playing file
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.553821 [DEBUG] switch_rtp.c:3280 RTP RECV DTMF 1:960
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.555825 [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154 Command Execute set(flex_digits)
>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/1000123402 at 192.168.0.154 set(flex_digits)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.555825 [DEBUG] mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154 SET [flex_digits]=[UNDEF]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [DEBUG] switch_channel.c:2560 (sofia/external/1000123402 at 192.168.0.154) Callstate Change ACTIVE -> HANGUP
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [NOTICE] sofia.c:538 Hangup sofia/external/1000123402 at 192.168.0.154 [CS_EXECUTE] [NORMAL_CLEARING]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [DEBUG] switch_channel.c:2576 Send signal sofia/external/1000123402 at 192.168.0.154 [KILL]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/1000123402 at 192.168.0.154 [BREAK]
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG] switch_core_session.c:2057 sofia/external/1000123402 at 192.168.0.154 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG] switch_cpp.cpp:988 sofia/external/1000123402 at 192.168.0.154 destroy/unlink session from object
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG] switch_core_session.c:2057 sofia/external/1000123402 at 192.168.0.154 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG] switch_core_codec.c:116 sofia/external/1000123402 at 192.168.0.154 Push codec L16:70
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246 [DEBUG] mod_conference.c:2815 Channel leaving conference, cause: NORMAL_CLEARING
>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.757239 [DEBUG] mod_conference.c:6104 sofia/external/1000123402 at 192.168.0.154 skip receive message [UNBRIDGE] (channel is hungup already)
>> 
>> I hope this info helps.
>> 
>> 2011/5/12 Michael Collins <msc at freeswitch.org>
>> 
>> 
>> On Thu, May 12, 2011 at 12:27 PM, Leonardo P. Bidinoto <leonardo.bidinoto at voicetechnology.com.br> wrote:
>> Hi Michael,
>> 
>> Im not using to any cdr module.
>> 
>> I would recommend that you do several things:
>> 
>> #1 - update to latest git
>> #2 - rotate logs
>> #3 - enable uuid logging (see logfile.conf.xml - param name "uuid")
>> #4 - reproduce the symptom with a single call (if possible)
>> #5 - pastebin the log for the uuid in question and link to it in this thread
>> 
>> From there hopefully we'll get a clue as to what is happening.
>> -MC
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> 
>> -- 
>> Leonardo Pires Bidinoto
>> Voice Technology
>> www.voicetechnology.com.br
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> 
>> -- 
>> Leonardo Pires Bidinoto
>> Voice Technology
>> www.voicetechnology.com.br
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>> http://www.freeswitch.org
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>> http://www.freeswitch.org
>> 
>> 
>> 
>> 
>> -- 
>> Leonardo Pires Bidinoto
>> Voice Technology
>> www.voicetechnology.com.br
>> 
>> 
>> 
>> -- 
>> Leonardo Pires Bidinoto
>> Voice Technology
>> www.voicetechnology.com.br
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Leonardo Pires Bidinoto
> Voice Technology
> www.voicetechnology.com.br
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From maciej.aniserowicz at gmail.com  Fri Jun  3 17:01:43 2011
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Fri, 3 Jun 2011 06:01:43 -0700 (PDT)
Subject: [Freeswitch-users] Eavesdrop between 2 FS instaces
In-Reply-To: <BANLkTinLdaHeaMy9XCthLM87jxQKNGSBXw@mail.gmail.com>
References: <1307039305975-6432366.post@n2.nabble.com>
	<BANLkTinLdaHeaMy9XCthLM87jxQKNGSBXw@mail.gmail.com>
Message-ID: <1307106103579-6435172.post@n2.nabble.com>

Hi Steve,
Thanks for interest.

Yes, i figured out that i cannot eavesdrop any 'easy' way. I tried the
approach you described (dialing into FS2 to a special context and passing
sip-h_X header). In fact i use this to bridge calls on different FSes. But
while i managed to get it working with bridging, i cannot find a way to do
eavesdropping.

With bridging I send:
uuid_transfer [CallA channel] bridge:{sip_h_X-target_channel_id=[CallB
channel]}sofia/bridging/123456 at ip:port,park inline

Then FS2 in dialplan for 123456 extracts target channel id from sip header
and intercepts it:
<condition field="destination_number" expression="^(123456)$">
    <action application="intercept" data="${sip_h_X-target_channel_id}" />
</condition>

What actions should i define in FS2 dialplan for eavesdropping? What inline
dialplan should i send in dial string? I'm stuck here.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Eavesdrop-between-2-FS-instaces-tp6432366p6435172.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From infos at madovsky.org  Fri Jun  3 17:54:22 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 3 Jun 2011 09:54:22 -0400
Subject: [Freeswitch-users] skype reverse engineered
References: <1307092839.6781.599.camel@luna.madrid.commsmundi.com><BANLkTikob7yF3ZP1yjB8QX9eTFC0EnfjzQ@mail.gmail.com><BANLkTiktSoB_qUWjWvyRrq3NFD3CUT6UWQ@mail.gmail.com>
	<1307104132721-6435062.post@n2.nabble.com>
Message-ID: <DD91BD0C144C45AE883DE7BB36E8C882@e1705>

I'm curious how MSSkype will manage
the open source code and community year after year....

----- Original Message ----- 
From: "mazilo" <Nabble at slickdeals.endjunk.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Friday, June 03, 2011 8:28 AM
Subject: Re: [Freeswitch-users] skype reverse engineered


>
> Steven Ayre wrote:
>> On the more optimistic note I have seen a good argument that there are
>> enough hardware Skype phones with Skype that they can't easily block 
>> older
>> versions without breaking a lot of devices.
> If Skype decides to block older version of software, then all it needs is 
> to
> furnish Skype Phone hardware manufacturers with a new code to come up with 
> a
> new firmware to support new Skype version. Then, hackers will start to
> reversed engineer on the new version. And, the cat/mouse chase begins.
>
> OTOH, it looks like the source was released to
> https://github.com/skypeopensource/skypeopensource/downloads GitHUB  on
> 6/2/11. let's see how fast it will become a Linux compilable
> package/plugin/module. I won't be surprised to see a working
> mod_skype_re(versed engineered) soon if Anthony and/or FS developers will
> decide to port the source into an FS module.
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 
> Watts of electricity.
> --
> View this message in context: 
> http://freeswitch-users.2379917.n2.nabble.com/skype-reverse-engineered-tp6434578p6435062.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From kahn at vestec.com  Fri Jun  3 05:48:01 2011
From: kahn at vestec.com (Kashif Kahn)
Date: Thu, 02 Jun 2011 21:48:01 -0400
Subject: [Freeswitch-users] Vestec ASR: 3 New Language Models
Message-ID: <4DE83D51.8020909@vestec.com>

Dear All,

We have launched three new acoustic models for use with Vestec ASR: (1) 
Australian English, (2) Chinese Mandarin, and (3) Indian English.

Vestec offers the best deal around for enabling sophisticated speech 
recognition with command-and-control type IVR applications. Our ASR 
supports SRGS grammar format, can be integrated using MRCP (v1 & v2) 
interface,  offers high recognition accuracy, and costs a fraction of 
conventional speech engines. We offer the ASR in two versions according 
to vocabulary size: 500 items vocabulary and 2,500 items vocabulary, per 
recognition. A starter kit is available for $25 from Vestec webstore: 
http://www.vestec.com/products

Please do not hesitate to contact me with any questions. We are looking 
forward to being of service.

Regards,
-Kashif

-- 
Kashif Kahn
VP Business Development
Vestec Inc
Waterloo, ON Canada
phone: +1 519 885-7615



From gopalakrishnan.an at gmail.com  Fri Jun  3 17:54:22 2011
From: gopalakrishnan.an at gmail.com (Gopal krishnan)
Date: Fri, 3 Jun 2011 19:24:22 +0530
Subject: [Freeswitch-users] skype reverse engineered
In-Reply-To: <1307104132721-6435062.post@n2.nabble.com>
References: <1307092839.6781.599.camel@luna.madrid.commsmundi.com>
	<BANLkTikob7yF3ZP1yjB8QX9eTFC0EnfjzQ@mail.gmail.com>
	<BANLkTiktSoB_qUWjWvyRrq3NFD3CUT6UWQ@mail.gmail.com>
	<1307104132721-6435062.post@n2.nabble.com>
Message-ID: <BANLkTimeNFKe56r3=JzPrw3zS0Z=HE99WA@mail.gmail.com>

http://www.zdnet.co.uk/news/intellectual-property/2011/06/03/skype-denounces-nefarious-reverse-engineering-40092982/

On Fri, Jun 3, 2011 at 5:58 PM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
> Steven Ayre wrote:
> > On the more optimistic note I have seen a good argument that there are
> > enough hardware Skype phones with Skype that they can't easily block
> older
> > versions without breaking a lot of devices.
> If Skype decides to block older version of software, then all it needs is
> to
> furnish Skype Phone hardware manufacturers with a new code to come up with
> a
> new firmware to support new Skype version. Then, hackers will start to
> reversed engineer on the new version. And, the cat/mouse chase begins.
>
> OTOH, it looks like the source was released to
> https://github.com/skypeopensource/skypeopensource/downloads GitHUB  on
> 6/2/11. let's see how fast it will become a Linux compilable
> package/plugin/module. I won't be surprised to see a working
> mod_skype_re(versed engineered) soon if Anthony and/or FS developers will
> decide to port the source into an FS module.
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3
> Watts of electricity.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/skype-reverse-engineered-tp6434578p6435062.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From andre.rosowski at omigos.de  Fri Jun  3 13:11:29 2011
From: andre.rosowski at omigos.de (=?ISO-8859-1?Q?Andr=E9_Rosowski?=)
Date: Fri, 03 Jun 2011 11:11:29 +0200
Subject: [Freeswitch-users] Any way to limit the ability to create
 conference rooms on demand?
In-Reply-To: <BANLkTikMwQePifbiVMPdcSg9nnEKUdG_Uw@mail.gmail.com>
References: <4DE3532A.9000004@omigos.de>
	<BANLkTikMwQePifbiVMPdcSg9nnEKUdG_Uw@mail.gmail.com>
Message-ID: <4DE8A541.6050807@omigos.de>

Hi Michael,

thank you for your answer. Could you be more specific how exactly I have 
to hard-code the conference rooms into the dialplan. Would you be so 
kind and give me an example? Much appreciated...


Regards

Realdoe

On 31.05.2011 22:58, Michael Collins wrote:
> There are only two practical ways to do this: hard-code your 
> conference rooms in the dialplan or use a database-backed solution 
> like mod_xml_curl. How are you currently managing the list of 
> available conferences?
>
> -MC
>
> 2011/5/30 Andr? Rosowski <andre.rosowski at omigos.de 
> <mailto:andre.rosowski at omigos.de>>
>
>     Hi there,
>
>     I'm using Bigbluebutton (online conferencing) with Freeswitch so that
>     users can call a number and then have to enter the "voicebridge"
>     number
>     to be put into a voice conference. The problem, however, is that if a
>     user dialed the wrong "voicebridge" number he will be put into a new
>     conference thus creating one on demand as stated in the describtion of
>     "mod_conference". Is there any way to manually create conference rooms
>     and make them static...not allowing for new conference "rooms" to be
>     created on demand.
>
>     default.xml in dialplan:
>
>     <extension name="phone conference">
>     <condition field="destination_number"
>     expression="^(SEND_TO_CONFERENCE)$">
>     <action application="conference" data="${pin}@wideband"/>
>     </condition>
>     </extension>
>
>     Thanks for your help : )
>
>     Regards,
>
>     Realdoe
>
>     --
>     Omigos Labs UG (haftungsbeschr?nkt) - Meisenstra?e 96 - 33607
>     Bielefeld
>     Fon: +49 521 2997 200 - Fax: +49 521 2997 101
>     info at omigos.de <mailto:info at omigos.de> - www.omigos.de
>     <http://www.omigos.de>
>     Gesch?ftsf?hrer: BSc Andr? Rosowski, Benjamin Bittner, Julian Klima
>     Registergericht: Amtsgericht Bielefeld - Registernummer: HRB 40255
>     USt-IdNr.: 305/5860/1585
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Omigos Labs UG (haftungsbeschr?nkt) - Meisenstra?e 96 - 33607 Bielefeld
Fon: +49 521 2997 200 - Fax: +49 521 2997 101
info at omigos.de - www.omigos.de
Gesch?ftsf?hrer: BSc Andr? Rosowski, Benjamin Bittner, Julian Klima
Registergericht: Amtsgericht Bielefeld - Registernummer: HRB 40255
USt-IdNr.: 305/5860/1585

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From nasir at ictinnovations.com  Fri Jun  3 18:12:04 2011
From: nasir at ictinnovations.com (Nasir Iqbal)
Date: Fri, 3 Jun 2011 19:12:04 +0500
Subject: [Freeswitch-users] Conference choppy voice
Message-ID: <BANLkTimbPVP3tqT=EuaZ0s7Y_9rSpWicRA@mail.gmail.com>

Hi,

I have configured a conference on new installation of freeswitch, every
thing works fine voice quality is good when lady says that "You are the only
one in this conference" and music but as soon as other party get connected
and started to talk voice get choppy/noisy nothing can be understood, like
if someone says "HELLO" other party listen jerking noisy
voice  like "---$H^----#E at ----%LL^----*O&"

I am new to Freeswitch please guide me that how I can fix
/ troubleshoot this issue or let me know if need some debugging etc...
currently my setup is following

CentOs-5 32bit
Intel(R) Pentium(R) D  CPU 2.66GHz

[root at localhost conf]# freeswitch -version
FreeSWITCH version: 1.0.head (git-66d16d1 2011-05-20 12-49-16 -0500)

[root at localhost conf]# uname -a
Linux localhost.localdomain 2.6.18-128.el5 #1 SMP Wed Jan 21 10:44:23 EST
2009 i686 i686 i386 GNU/Linux

[root at localhost conf]# free
             total       used       free     shared    buffers     cached
Mem:       1001664     959668      41996          0     196484     540676
-/+ buffers/cache:     222508     779156
Swap:      2031608         60    2031548


<?xml version="1.0"?>
<include>
  <configuration name="conference.conf">
    <caller-controls>
      <group name="default-keymap">
        <control action="vol talk dn" digits="1"/>
        <control action="vol talk zero" digits="2"/>
        <control action="vol talk up" digits="3"/>
        <control action="vol listen dn" digits="4"/>
        <control action="vol listen zero" digits="5"/>
        <control action="vol listen up" digits="6"/>
        <control action="energy dn" digits="7"/>
        <control action="energy equ" digits="8"/>
        <control action="energy up" digits="9"/>
        <control action="deaf mute" digits="*"/>
        <control action="mute" digits="0"/>
      </group>
    </caller-controls>
    <profiles>
      <profile name="conference_1">
        <param name="rate" value="8000"/> <!-- Here I tried all possible
values but no luck -->
        <param name="interval" value="20"/>
        <param name="energy-level" value="20"/>
        <param name="sound-prefix" value="$${sounds_dir}/en/us/callie"/>
        <param name="caller-controls" value="default-keymap"/>
        <param name="tts-engine" value="flite"/>
        <param name="tts-voice" value="kal"/>
        <param name="muted-sound" value="conference/conf-muted.wav"/>
        <param name="mute-detect-sound" value=""/>
        <param name="max-members" value=""/>
        <param name="max-members-sound" value=""/>
        <param name="comfort-noise" value="1"/>
        <param name="suppress-events" value=""/>
        <param name="verbose-events" value=""/>
        <param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
        <param name="alone-sound" value="conference/conf-alone.wav"/>
        <param name="moh-sound" value="local_stream://moh"/>
        <param name="enter-sound"
value="tone_stream://%(200,0,500,600,700)"/>
        <param name="exit-sound"
value="tone_stream://%(500,0,300,200,100,50,25)"/>
        <param name="kicked-sound" value="conference/conf-kicked.wav"/>
        <param name="locked-sound" value="conference/conf-locked.wav"/>
        <param name="is-locked-sound"
value="conference/conf-is-locked.wav"/>
        <param name="is-unlocked-sound"
value="conference/conf-is-unlocked.wav"/>
        <param name="pin-sound" value="conference/conf-pin.wav"/>
        <param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
      </profile>
    </profiles>
  </configuration>
</include>


Thanks in advance

Nasir
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From infos at madovsky.org  Fri Jun  3 19:02:01 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 3 Jun 2011 11:02:01 -0400
Subject: [Freeswitch-users] skype reverse engineered
References: <1307092839.6781.599.camel@luna.madrid.commsmundi.com><BANLkTikob7yF3ZP1yjB8QX9eTFC0EnfjzQ@mail.gmail.com><BANLkTiktSoB_qUWjWvyRrq3NFD3CUT6UWQ@mail.gmail.com><1307104132721-6435062.post@n2.nabble.com>
	<BANLkTimeNFKe56r3=JzPrw3zS0Z=HE99WA@mail.gmail.com>
Message-ID: <A8F4D3FF0E3748D2B62FA356A4165A68@e1705>

some logical I don't understand.
why is it need reverse engineering if Skype had an open source community until today ?
  ----- Original Message ----- 
  From: Gopal krishnan 
  To: FreeSWITCH Users Help 
  Sent: Friday, June 03, 2011 9:54 AM
  Subject: Re: [Freeswitch-users] skype reverse engineered


  http://www.zdnet.co.uk/news/intellectual-property/2011/06/03/skype-denounces-nefarious-reverse-engineering-40092982/


  On Fri, Jun 3, 2011 at 5:58 PM, mazilo <Nabble at slickdeals.endjunk.com> wrote:


    Steven Ayre wrote:
    > On the more optimistic note I have seen a good argument that there are
    > enough hardware Skype phones with Skype that they can't easily block older
    > versions without breaking a lot of devices.

    If Skype decides to block older version of software, then all it needs is to
    furnish Skype Phone hardware manufacturers with a new code to come up with a
    new firmware to support new Skype version. Then, hackers will start to
    reversed engineer on the new version. And, the cat/mouse chase begins.

    OTOH, it looks like the source was released to
    https://github.com/skypeopensource/skypeopensource/downloads GitHUB  on
    6/2/11. let's see how fast it will become a Linux compilable
    package/plugin/module. I won't be surprised to see a working
    mod_skype_re(versed engineered) soon if Anthony and/or FS developers will
    decide to port the source into an FS module.

    -----
    FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
    --
    View this message in context: http://freeswitch-users.2379917.n2.nabble.com/skype-reverse-engineered-tp6434578p6435062.html
    Sent from the freeswitch-users mailing list archive at Nabble.com.


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





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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From steveayre at gmail.com  Fri Jun  3 19:08:03 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 3 Jun 2011 16:08:03 +0100
Subject: [Freeswitch-users] skype reverse engineered
In-Reply-To: <A8F4D3FF0E3748D2B62FA356A4165A68@e1705>
References: <1307092839.6781.599.camel@luna.madrid.commsmundi.com>
	<BANLkTikob7yF3ZP1yjB8QX9eTFC0EnfjzQ@mail.gmail.com>
	<BANLkTiktSoB_qUWjWvyRrq3NFD3CUT6UWQ@mail.gmail.com>
	<1307104132721-6435062.post@n2.nabble.com>
	<BANLkTimeNFKe56r3=JzPrw3zS0Z=HE99WA@mail.gmail.com>
	<A8F4D3FF0E3748D2B62FA356A4165A68@e1705>
Message-ID: <BANLkTi=Sb3cimEQOc6Knyc5j5Jyhty3LfQ@mail.gmail.com>

There's nothing open about about Skype. The protocol is entirely closed and
proprietary.

They offer an API to talk to the Skype client from another program running
on the same machine, but that's about it. mod_skypopen works by running one
or more copies of the official Skype client, creating a virtual sound card
and communicating with that from FreeSWITCH.

-Steve



On 3 June 2011 16:02, Madovsky <infos at madovsky.org> wrote:

>  some logical I don't understand.
> why is it need reverse engineering if Skype had an open source community
> until today ?
>
> ----- Original Message -----
> *From:* Gopal krishnan <gopalakrishnan.an at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Friday, June 03, 2011 9:54 AMS
> *Subject:* Re: [Freeswitch-users] skype reverse engineered
>
>
> http://www.zdnet.co.uk/news/intellectual-property/2011/06/03/skype-denounces-nefarious-reverse-engineering-40092982/
>
> On Fri, Jun 3, 2011 at 5:58 PM, mazilo <Nabble at slickdeals.endjunk.com>wrote:
>
>>
>> Steven Ayre wrote:
>> > On the more optimistic note I have seen a good argument that there are
>> > enough hardware Skype phones with Skype that they can't easily block
>> older
>> > versions without breaking a lot of devices.
>> If Skype decides to block older version of software, then all it needs is
>> to
>> furnish Skype Phone hardware manufacturers with a new code to come up with
>> a
>> new firmware to support new Skype version. Then, hackers will start to
>> reversed engineer on the new version. And, the cat/mouse chase begins.
>>
>> OTOH, it looks like the source was released to
>> https://github.com/skypeopensource/skypeopensource/downloads GitHUB  on
>> 6/2/11. let's see how fast it will become a Linux compilable
>> package/plugin/module. I won't be surprised to see a working
>> mod_skype_re(versed engineered) soon if Anthony and/or FS developers will
>> decide to port the source into an FS module.
>>
>> -----
>> FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3
>> Watts of electricity.
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/skype-reverse-engineered-tp6434578p6435062.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Fri Jun  3 19:11:53 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 3 Jun 2011 16:11:53 +0100
Subject: [Freeswitch-users] Eavesdrop between 2 FS instaces
In-Reply-To: <1307106103579-6435172.post@n2.nabble.com>
References: <1307039305975-6432366.post@n2.nabble.com>
	<BANLkTinLdaHeaMy9XCthLM87jxQKNGSBXw@mail.gmail.com>
	<1307106103579-6435172.post@n2.nabble.com>
Message-ID: <BANLkTi=aMvSa9Ey=hzXzoAsJEGNqZcpbCg@mail.gmail.com>

I think you want the eavesdrop app:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop

Intercept takes over the call. Eavesdrop just silently listens in.

-Steve



On 3 June 2011 14:01, Maciej Aniserowicz <maciej.aniserowicz at gmail.com>wrote:

> Hi Steve,
> Thanks for interest.
>
> Yes, i figured out that i cannot eavesdrop any 'easy' way. I tried the
> approach you described (dialing into FS2 to a special context and passing
> sip-h_X header). In fact i use this to bridge calls on different FSes. But
> while i managed to get it working with bridging, i cannot find a way to do
> eavesdropping.
>
> With bridging I send:
> uuid_transfer [CallA channel] bridge:{sip_h_X-target_channel_id=[CallB
> channel]}sofia/bridging/123456 at ip:port,park inline
>
> Then FS2 in dialplan for 123456 extracts target channel id from sip header
> and intercepts it:
> <condition field="destination_number" expression="^(123456)$">
>    <action application="intercept" data="${sip_h_X-target_channel_id}" />
> </condition>
>
> What actions should i define in FS2 dialplan for eavesdropping? What inline
> dialplan should i send in dial string? I'm stuck here.
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Eavesdrop-between-2-FS-instaces-tp6432366p6435172.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From peter.olsson at visionutveckling.se  Fri Jun  3 19:13:59 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 3 Jun 2011 17:13:59 +0200
Subject: [Freeswitch-users] Conference choppy voice
In-Reply-To: <BANLkTimbPVP3tqT=EuaZ0s7Y_9rSpWicRA@mail.gmail.com>
References: <BANLkTimbPVP3tqT=EuaZ0s7Y_9rSpWicRA@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59DE2B6327@cooper>

First check the comfort-noise value - default is 1400, try that.

And also (if the above doesn't help) - do a timer_test in the FS console and get back with the results.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Nasir Iqbal [nasir at ictinnovations.com]
Skickat: den 3 juni 2011 16:12
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] Conference choppy voice

Hi,

I have configured a conference on new installation of freeswitch, every thing works fine voice quality is good when lady says that "You are the only one in this conference" and music but as soon as other party get connected and started to talk voice get choppy/noisy nothing can be understood, like if someone says "HELLO" other party listen jerking noisy voice  like "---$H^----#E at ----%LL^----*O&"

I am new to Freeswitch please guide me that how I can fix / troubleshoot this issue or let me know if need some debugging etc... currently my setup is following

CentOs-5 32bit
Intel(R) Pentium(R) D  CPU 2.66GHz

[root at localhost conf]# freeswitch -version
FreeSWITCH version: 1.0.head (git-66d16d1 2011-05-20 12-49-16 -0500)

[root at localhost conf]# uname -a
Linux localhost.localdomain 2.6.18-128.el5 #1 SMP Wed Jan 21 10:44:23 EST 2009 i686 i686 i386 GNU/Linux

[root at localhost conf]# free
             total       used       free     shared    buffers     cached
Mem:       1001664     959668      41996          0     196484     540676
-/+ buffers/cache:     222508     779156
Swap:      2031608         60    2031548


<?xml version="1.0"?>
<include>
  <configuration name="conference.conf">
    <caller-controls>
      <group name="default-keymap">
        <control action="vol talk dn" digits="1"/>
        <control action="vol talk zero" digits="2"/>
        <control action="vol talk up" digits="3"/>
        <control action="vol listen dn" digits="4"/>
        <control action="vol listen zero" digits="5"/>
        <control action="vol listen up" digits="6"/>
        <control action="energy dn" digits="7"/>
        <control action="energy equ" digits="8"/>
        <control action="energy up" digits="9"/>
        <control action="deaf mute" digits="*"/>
        <control action="mute" digits="0"/>
      </group>
    </caller-controls>
    <profiles>
      <profile name="conference_1">
        <param name="rate" value="8000"/> <!-- Here I tried all possible values but no luck -->
        <param name="interval" value="20"/>
        <param name="energy-level" value="20"/>
        <param name="sound-prefix" value="$${sounds_dir}/en/us/callie"/>
        <param name="caller-controls" value="default-keymap"/>
        <param name="tts-engine" value="flite"/>
        <param name="tts-voice" value="kal"/>
        <param name="muted-sound" value="conference/conf-muted.wav"/>
        <param name="mute-detect-sound" value=""/>
        <param name="max-members" value=""/>
        <param name="max-members-sound" value=""/>
        <param name="comfort-noise" value="1"/>
        <param name="suppress-events" value=""/>
        <param name="verbose-events" value=""/>
        <param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
        <param name="alone-sound" value="conference/conf-alone.wav"/>
        <param name="moh-sound" value="local_stream://moh"/>
        <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
        <param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
        <param name="kicked-sound" value="conference/conf-kicked.wav"/>
        <param name="locked-sound" value="conference/conf-locked.wav"/>
        <param name="is-locked-sound" value="conference/conf-is-locked.wav"/>
        <param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/>
        <param name="pin-sound" value="conference/conf-pin.wav"/>
        <param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
      </profile>
    </profiles>
  </configuration>
</include>


Thanks in advance

Nasir
!DSPAM:4de8ec6d32764556817595!


From infos at madovsky.org  Fri Jun  3 19:20:12 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 3 Jun 2011 11:20:12 -0400
Subject: [Freeswitch-users] skype reverse engineered
References: <1307092839.6781.599.camel@luna.madrid.commsmundi.com><BANLkTikob7yF3ZP1yjB8QX9eTFC0EnfjzQ@mail.gmail.com><BANLkTiktSoB_qUWjWvyRrq3NFD3CUT6UWQ@mail.gmail.com><1307104132721-6435062.post@n2.nabble.com><BANLkTimeNFKe56r3=JzPrw3zS0Z=HE99WA@mail.gmail.com><A8F4D3FF0E3748D2B62FA356A4165A68@e1705>
	<BANLkTi=Sb3cimEQOc6Knyc5j5Jyhty3LfQ@mail.gmail.com>
Message-ID: <30241955FD5E4862A155DDD8E8BE8179@e1705>

tha'ts clear, not sure it will be a good future for them so....
  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Friday, June 03, 2011 11:08 AM
  Subject: Re: [Freeswitch-users] skype reverse engineered


  There's nothing open about about Skype. The protocol is entirely closed and proprietary.

  They offer an API to talk to the Skype client from another program running on the same machine, but that's about it. mod_skypopen works by running one or more copies of the official Skype client, creating a virtual sound card and communicating with that from FreeSWITCH.

  -Steve




  On 3 June 2011 16:02, Madovsky <infos at madovsky.org> wrote:

    some logical I don't understand.
    why is it need reverse engineering if Skype had an open source community until today ?
      ----- Original Message ----- 
      From: Gopal krishnan 
      To: FreeSWITCH Users Help 
      Sent: Friday, June 03, 2011 9:54 AMS
      Subject: Re: [Freeswitch-users] skype reverse engineered


      http://www.zdnet.co.uk/news/intellectual-property/2011/06/03/skype-denounces-nefarious-reverse-engineering-40092982/


      On Fri, Jun 3, 2011 at 5:58 PM, mazilo <Nabble at slickdeals.endjunk.com> wrote:


        Steven Ayre wrote:
        > On the more optimistic note I have seen a good argument that there are
        > enough hardware Skype phones with Skype that they can't easily block older
        > versions without breaking a lot of devices.

        If Skype decides to block older version of software, then all it needs is to
        furnish Skype Phone hardware manufacturers with a new code to come up with a
        new firmware to support new Skype version. Then, hackers will start to
        reversed engineer on the new version. And, the cat/mouse chase begins.

        OTOH, it looks like the source was released to
        https://github.com/skypeopensource/skypeopensource/downloads GitHUB  on
        6/2/11. let's see how fast it will become a Linux compilable
        package/plugin/module. I won't be surprised to see a working
        mod_skype_re(versed engineered) soon if Anthony and/or FS developers will
        decide to port the source into an FS module.

        -----
        FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
        --
        View this message in context: http://freeswitch-users.2379917.n2.nabble.com/skype-reverse-engineered-tp6434578p6435062.html
        Sent from the freeswitch-users mailing list archive at Nabble.com.


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





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


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      http://www.freeswitch.org



    _______________________________________________
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    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From david.ponzone at ipeva.fr  Fri Jun  3 19:22:23 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 3 Jun 2011 17:22:23 +0200
Subject: [Freeswitch-users] Conference choppy voice
In-Reply-To: <BANLkTimbPVP3tqT=EuaZ0s7Y_9rSpWicRA@mail.gmail.com>
References: <BANLkTimbPVP3tqT=EuaZ0s7Y_9rSpWicRA@mail.gmail.com>
Message-ID: <BDF6EE3E-AB7D-4D4A-81E6-74FED71EB439@ipeva.fr>

Check the timer precision in your kernel.
It should be 1000Hz.

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

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

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




Le 03/06/2011 ? 16:12, Nasir Iqbal a ?crit :

> Hi,
> 
> I have configured a conference on new installation of freeswitch, every thing works fine voice quality is good when lady says that "You are the only one in this conference" and music but as soon as other party get connected and started to talk voice get choppy/noisy nothing can be understood, like if someone says "HELLO" other party listen jerking noisy voice  like "---$H^----#E at ----%LL^----*O&"
> 
> I am new to Freeswitch please guide me that how I can fix / troubleshoot this issue or let me know if need some debugging etc... currently my setup is following
> 
> CentOs-5 32bit
> Intel(R) Pentium(R) D  CPU 2.66GHz
> 
> [root at localhost conf]# freeswitch -version
> FreeSWITCH version: 1.0.head (git-66d16d1 2011-05-20 12-49-16 -0500)
> 
> [root at localhost conf]# uname -a
> Linux localhost.localdomain 2.6.18-128.el5 #1 SMP Wed Jan 21 10:44:23 EST 2009 i686 i686 i386 GNU/Linux
> 
> [root at localhost conf]# free
>              total       used       free     shared    buffers     cached
> Mem:       1001664     959668      41996          0     196484     540676
> -/+ buffers/cache:     222508     779156
> Swap:      2031608         60    2031548
> 
> 
> <?xml version="1.0"?>
> <include>
>   <configuration name="conference.conf">
>     <caller-controls>
>       <group name="default-keymap">
>         <control action="vol talk dn" digits="1"/>
>         <control action="vol talk zero" digits="2"/>
>         <control action="vol talk up" digits="3"/>
>         <control action="vol listen dn" digits="4"/>
>         <control action="vol listen zero" digits="5"/>
>         <control action="vol listen up" digits="6"/>
>         <control action="energy dn" digits="7"/>
>         <control action="energy equ" digits="8"/>
>         <control action="energy up" digits="9"/>
>         <control action="deaf mute" digits="*"/>
>         <control action="mute" digits="0"/>
>       </group>
>     </caller-controls>
>     <profiles>
>       <profile name="conference_1">
>         <param name="rate" value="8000"/> <!-- Here I tried all possible values but no luck -->
>         <param name="interval" value="20"/>
>         <param name="energy-level" value="20"/>
>         <param name="sound-prefix" value="$${sounds_dir}/en/us/callie"/>
>         <param name="caller-controls" value="default-keymap"/>
>         <param name="tts-engine" value="flite"/>
>         <param name="tts-voice" value="kal"/>
>         <param name="muted-sound" value="conference/conf-muted.wav"/>
>         <param name="mute-detect-sound" value=""/>
>         <param name="max-members" value=""/>
>         <param name="max-members-sound" value=""/>
>         <param name="comfort-noise" value="1"/>
>         <param name="suppress-events" value=""/>
>         <param name="verbose-events" value=""/>
>         <param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
>         <param name="alone-sound" value="conference/conf-alone.wav"/>
>         <param name="moh-sound" value="local_stream://moh"/>
>         <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
>         <param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
>         <param name="kicked-sound" value="conference/conf-kicked.wav"/>
>         <param name="locked-sound" value="conference/conf-locked.wav"/>
>         <param name="is-locked-sound" value="conference/conf-is-locked.wav"/>
>         <param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/>
>         <param name="pin-sound" value="conference/conf-pin.wav"/>
>         <param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
>       </profile>
>     </profiles>
>   </configuration>
> </include>
> 
> 
> Thanks in advance
> 
> Nasir
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From john_platts at hotmail.com  Fri Jun  3 19:33:33 2011
From: john_platts at hotmail.com (John Platts)
Date: Fri, 3 Jun 2011 10:33:33 -0500
Subject: [Freeswitch-users] Click-to-call from Outlook
Message-ID: <SNT116-W414898E6E31CC59968CBA79D7F0@phx.gbl>


Is there already an application that allows calls to be initiated from FreeSWITCH via Microsoft Outlook? We really want to initiate calls from FreeSWITCH through Microsoft Outlook?
 		 	   		  
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From curriegrad2004 at gmail.com  Fri Jun  3 19:45:23 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Fri, 3 Jun 2011 08:45:23 -0700
Subject: [Freeswitch-users] Click-to-call from Outlook
In-Reply-To: <SNT116-W414898E6E31CC59968CBA79D7F0@phx.gbl>
References: <SNT116-W414898E6E31CC59968CBA79D7F0@phx.gbl>
Message-ID: <BANLkTinYKWSmJgXLWWLAArNsfP_-vwqPZw@mail.gmail.com>

If outlook does support the sip: uri, then it should be able to
automatically have a function similar to what you are looking for.
Clicking on a SIP URI should launch a softphone where you should be
able to dial out to it.

On Fri, Jun 3, 2011 at 8:33 AM, John Platts <john_platts at hotmail.com> wrote:
> Is there already an application that allows calls to be initiated from
> FreeSWITCH via Microsoft Outlook? We really want to initiate calls from
> FreeSWITCH through Microsoft Outlook?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From avi at avimarcus.net  Fri Jun  3 20:01:03 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 3 Jun 2011 19:01:03 +0300
Subject: [Freeswitch-users] Click-to-call from Outlook
In-Reply-To: <BANLkTinYKWSmJgXLWWLAArNsfP_-vwqPZw@mail.gmail.com>
References: <SNT116-W414898E6E31CC59968CBA79D7F0@phx.gbl>
	<BANLkTinYKWSmJgXLWWLAArNsfP_-vwqPZw@mail.gmail.com>
Message-ID: <BANLkTi=-WM_tp8qhiGBX86MbTndv1hz4wg@mail.gmail.com>

Or, if you want to click a button outlook to ring your deskphone and call
the contact, you can set up a web url that use the api calls to run
"originates".
-Avi

On Fri, Jun 3, 2011 at 6:45 PM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:

> If outlook does support the sip: uri, then it should be able to
> automatically have a function similar to what you are looking for.
> Clicking on a SIP URI should launch a softphone where you should be
> able to dial out to it.
>
> On Fri, Jun 3, 2011 at 8:33 AM, John Platts <john_platts at hotmail.com>
> wrote:
> > Is there already an application that allows calls to be initiated from
> > FreeSWITCH via Microsoft Outlook? We really want to initiate calls from
> > FreeSWITCH through Microsoft Outlook?
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From b_ball_henry at hotmail.com  Fri Jun  3 20:19:53 2011
From: b_ball_henry at hotmail.com (Henry Huang)
Date: Sat, 4 Jun 2011 00:19:53 +0800
Subject: [Freeswitch-users] mod_skypopen is awesome issue with demo-ivr
	conference
Message-ID: <BANLkTik8hpzwjo-Susi6-wFDAzPiDZ-Jyw@mail.gmail.com>

Hi:

The new interactive installation for mod_skypopen is a piece of cake. Made
live much easier. I was so excited and tried it out right after the
installation was done. But I found a few issues calling the demo-ivr from
skyTpe client.

1. The screaming monkey does not scream anymore.
2. The conference that connects to the freeswitch.org doesn't respond to
dtmf pressed on the skype client. Ever since the second client is connected
to the conference, the conference tell the client it's muted. So I tried
push 0 and any other key to see if I get to unmute. But no, non of the key
press works.

If anyone have previous experiences on these and fixed them, please do share
your methods.

Sound quality with the new virtual sound drive is very good.

Thanks,

Henry Huang
US: +1(818)6885508 | ??(Taiwan): +886 933847619
Contact Me [image:
LinkedIn]<http://www.linkedin.com/profile/view?id=4590380> [image:
Facebook] <http://www.facebook.com/profile.php?id=595148342> [image:
Twitter] <http://twitter.com/unicsolution>
IM [image: Google Talk] red_rain_seven at gmail.com [image: Skype] unicsolution
[image: MSN] b_ball_henry at hotmail.com
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From msc at freeswitch.org  Fri Jun  3 20:35:46 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 3 Jun 2011 09:35:46 -0700
Subject: [Freeswitch-users] mod_skypopen is awesome issue with demo-ivr
	conference
In-Reply-To: <BANLkTik8hpzwjo-Susi6-wFDAzPiDZ-Jyw@mail.gmail.com>
References: <BANLkTik8hpzwjo-Susi6-wFDAzPiDZ-Jyw@mail.gmail.com>
Message-ID: <BANLkTimu3GAitdDLvk=tcU+ZYUuXAjKCrw@mail.gmail.com>

The monkeys have stopped screaming for everyone. :( I have been looking for
an alternative sound file for this but haven't found anything I really like.

Not sure about the DTMF, but from what I understand there have been issues
with sending DTMFs from the Skype client. I haven't tried it myself so I
will defer to those who have.

-MC

On Fri, Jun 3, 2011 at 9:19 AM, Henry Huang <b_ball_henry at hotmail.com>wrote:

> Hi:
>
> The new interactive installation for mod_skypopen is a piece of cake. Made
> live much easier. I was so excited and tried it out right after the
> installation was done. But I found a few issues calling the demo-ivr from
> skyTpe client.
>
> 1. The screaming monkey does not scream anymore.
> 2. The conference that connects to the freeswitch.org doesn't respond to
> dtmf pressed on the skype client. Ever since the second client is connected
> to the conference, the conference tell the client it's muted. So I tried
> push 0 and any other key to see if I get to unmute. But no, non of the key
> press works.
>
> If anyone have previous experiences on these and fixed them, please do
> share your methods.
>
> Sound quality with the new virtual sound drive is very good.
>
> Thanks,
>
> Henry Huang
> US: +1(818)6885508 | ??(Taiwan): +886 933847619
> Contact Me [image: LinkedIn]<http://www.linkedin.com/profile/view?id=4590380> [image:
> Facebook] <http://www.facebook.com/profile.php?id=595148342> [image:
> Twitter] <http://twitter.com/unicsolution>
> IM [image: Google Talk] red_rain_seven at gmail.com [image: Skype]unicsolution [image:
> MSN] b_ball_henry at hotmail.com
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Fri Jun  3 20:37:55 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 3 Jun 2011 11:37:55 -0500
Subject: [Freeswitch-users] Any way to limit the ability to create
 conference rooms on demand?
In-Reply-To: <4DE8A541.6050807@omigos.de>
References: <4DE3532A.9000004@omigos.de>
	<BANLkTikMwQePifbiVMPdcSg9nnEKUdG_Uw@mail.gmail.com>
	<4DE8A541.6050807@omigos.de>
Message-ID: <BANLkTikriZf53neyX-31kjQRiUeHHOV3Rg@mail.gmail.com>

<extension name="phone conference">
<condition field="destination_number" expression="^(SEND_TO_CONFERENCE)$"/>
<condition field="${pin}" expression="5\d{3}"/>
<action application="conference" data="${pin}@wideband"/>
<anti-action application="playback" data="invalidpin.wav"/>
</condition>
</extension>

would limit you to pins that were 5xxx


2011/6/3 Andr? Rosowski <andre.rosowski at omigos.de>:
> Hi Michael,
>
> thank you for your answer. Could you be more specific how exactly I have to
> hard-code the conference rooms into the dialplan. Would you be so kind and
> give me an example? Much appreciated...
>
>
> Regards
>
> Realdoe
>
> On 31.05.2011 22:58, Michael Collins wrote:
>
> There are only two practical ways to do this: hard-code your conference
> rooms in the dialplan or use a database-backed solution like mod_xml_curl.
> How are you currently managing the list of available conferences?
> -MC
>
> 2011/5/30 Andr? Rosowski <andre.rosowski at omigos.de>
>>
>> Hi there,
>>
>> I'm using Bigbluebutton (online conferencing) with Freeswitch so that
>> users can call a number and then have to enter the "voicebridge" number
>> to be put into a voice conference. The problem, however, is that if a
>> user dialed the wrong "voicebridge" number he will be put into a new
>> conference thus creating one on demand as stated in the describtion of
>> "mod_conference". Is there any way to manually create conference rooms
>> and make them static...not allowing for new conference "rooms" to be
>> created on demand.
>>
>> default.xml in dialplan:
>>
>> <extension name="phone conference">
>> <condition field="destination_number" expression="^(SEND_TO_CONFERENCE)$">
>> <action application="conference" data="${pin}@wideband"/>
>> </condition>
>> </extension>
>>
>> Thanks for your help : )
>>
>> Regards,
>>
>> Realdoe
>>
>> --
>> Omigos Labs UG (haftungsbeschr?nkt) - Meisenstra?e 96 - 33607 Bielefeld
>> Fon: +49 521 2997 200 - Fax: +49 521 2997 101
>> info at omigos.de - www.omigos.de
>> Gesch?ftsf?hrer: BSc Andr? Rosowski, Benjamin Bittner, Julian Klima
>> Registergericht: Amtsgericht Bielefeld - Registernummer: HRB 40255
>> USt-IdNr.: 305/5860/1585
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> --
> Omigos Labs UG (haftungsbeschr?nkt) - Meisenstra?e 96 - 33607 Bielefeld
> Fon: +49 521 2997 200 - Fax: +49 521 2997 101
> info at omigos.de - www.omigos.de
> Gesch?ftsf?hrer: BSc Andr? Rosowski, Benjamin Bittner, Julian Klima
> Registergericht: Amtsgericht Bielefeld - Registernummer: HRB 40255
> USt-IdNr.: 305/5860/1585
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From msc at freeswitch.org  Fri Jun  3 20:44:35 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 3 Jun 2011 09:44:35 -0700
Subject: [Freeswitch-users] Transferring to a cell phone
In-Reply-To: <33095823FD21DF429B481B5163264B7950AC4FB5F8@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950AC3AC74C@VMBX102.ihostexchange.net>
	<0454EB31-E9EA-4594-9FDD-29D69F5B6126@ipeva.fr>
	<33095823FD21DF429B481B5163264B7950AC3AC7E2@VMBX102.ihostexchange.net>
	<1306499299750-6410887.post@n2.nabble.com>
	<33095823FD21DF429B481B5163264B7950AC3AC8B0@VMBX102.ihostexchange.net>
	<BANLkTikrgnaTDL9+WtXh_4RAWTFnmja71Q@mail.gmail.com>
	<33095823FD21DF429B481B5163264B7950AC3AC938@VMBX102.ihostexchange.net>
	<33095823FD21DF429B481B5163264B7950AC3ACA1B@VMBX102.ihostexchange.net>
	<BANLkTi=4fMHUkHjoCZ8exuw6Qgg5Pgj6Cw@mail.gmail.com>
	<33095823FD21DF429B481B5163264B7950AC4FB5F3@VMBX102.ihostexchange.net>
	<BANLkTimesjA5yzOFT+oVWSZzLPDVXZofrw@mail.gmail.com>
	<33095823FD21DF429B481B5163264B7950AC4FB5F8@VMBX102.ihostexchange.net>
Message-ID: <BANLkTinEULJpZn8jX=h70UW8PY6UUNVWwQ@mail.gmail.com>

Why are you using js to begin with? You don't really gain anything by
performing the transfer from within a js script. In fact, when you do it
this way the js instance stays active for the entire call. A happy medium
might be to session.execute("transfer","2223334444 XML custom"); and then
have a custom dp context handle the bridge. That way you could do whatever
it is you need to do inside of js and still let the dialplan do its thing.

-MC

On Wed, Jun 1, 2011 at 5:17 PM, Yungwei Chen <yungwei at resolvity.com> wrote:

> Using bridge in the dialplan directly works.
>
> I tried putting only the following line in transfer.js, but it doesn't
> work.
>       session.execute("bridge", "sofia/gateway/broadvoice/2223334444");
>
> Putting the following instead doesn't work either.
>       session.execute("bridge",
> "{ignore_early_media=true}sofia/gateway/broadvoice/2223334444");
>
> Using the following instead doesn't work either.
>       session.answer();
>       session.execute("bridge",
> "{ignore_early_media=true}sofia/gateway/broadvoice/2223334444");
>
> Still don't understand why.
>
> From: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
> Collins [msc at freeswitch.org]
> Sent: Wednesday, June 01, 2011 4:32 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Transferring to a cell phone
>
>
> I'm not really sure what's happening here. The first BYE message that I see
> comes in at line #548 from the provider side of what I assume is the A leg.
>
>
> For now I would try using a very simple bridge app right in the public
> context and avoid all the javascript so that you can get the simplest
> scenario. SImply route the A leg to the bridge app and bridge it to the
> broadvoice gateway:
> <extension name="route DID">
>  <condition field="destination_number"
> expression="^(INCOMING_DID_NUMBER)$">
>    <action application="bridge"
> data="sofia/gateway/broadvoice/2223334444"/>
>  </condition>
> </extension>
>
>
>
> A call coming in from one SIP endpoint and being bridged to another SIP
> endpoint is pretty much the core and fundamental functionality of
> FreeSWITCH, so I'm thinking that the javascript is doing something
> unintended.
>
>
> -MC
>
>
> On Wed, Jun 1, 2011 at 9:52 AM, Yungwei Chen <yungwei at resolvity.com>
> wrote:
>
> Here's the log with siptrace.
> http://pastebin.freeswitch.org/16424
>
> Here's the content of transfer.js:
>   session.answer();
>   if (session.ready())
>   {
>       session.execute("bridge",
> "{ignore_early_media=true}sofia/gateway/broadvoice/2223334444");
>   }
>
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
> Collins [msc at freeswitch.org]
> Sent: Friday, May 27, 2011 6:21 PM
>
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Transferring to a cell phone
>
>
>
> Something is not right with the outbound leg of the call. It looks like
> there is an immediate hangup after the b leg answers. Get a siptrace of that
> traffic and look to see what is causing the hangup.
>
>
> -MC
>
>
> On Fri, May 27, 2011 at 2:27 PM, Yungwei Chen <yungwei at resolvity.com>
> wrote:
>
> The whole picture looks like the following:
> A land-line<-->SIP provider A<-->FS<-->SIP provider A<-->a cell phone
>
> Is there any configuration setting that needs to be enabled or set
> differently so that those 2 endpoints can talk to each other?
> Btw, the same scenario works with Asterisk so my SIP provider shouldn't be
> the problem.
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yungwei Chen
> Sent: Friday, May 27, 2011 11:34 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Transferring to a cell phone
>
> Here's the debug log, http://pastebin.freeswitch.org/16398
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
> Collins
> Sent: Friday, May 27, 2011 10:08 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Transferring to a cell phone
>
> You need to put a debug log on pastebin.freeswitch.org. Be sure to use
> "FreeSWITCH Log" for the syntax highlighting.
> -MC
> On Fri, May 27, 2011 at 7:41 AM, Yungwei Chen <yungwei at resolvity.com>
> wrote:
> No.
>
> I just verified that making an outbound call to my cell phone still works.
> I even recorded the session just to make sure, and in the recording I hear
> things both ways.
> I first dial 9911 from my SIP client (behind freeswitch), and this leads to
> the javascript program below.
>  session.answer();
>  var new_session = new
> Session("{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");
>  session.hangup(16); // disconnects the session between the SIP client and
> freeswitch
>  new_session.execute("record_session", "/tmp/call-from-js-to-cell.wav");
>  while (new_session.ready())
>  {
>      new_session.streamFile("/path/to/local/wav/file");
>  }
>
> Now back to the case I'm having problem with.
> In this case, I first make a call from a landline to freeswitch through my
> sip provider, and then a javascript program takes over.
> I want to transfer the call to a cell phone so that the landline and the
> cell phone can communicate with each other.
> Here's the javascript program:
>  session.answer();
>
>  if (session.ready())
>  {
>      session.execute("bridge",
> "{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");
>  }
>
> So what am I missing here?
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of mazilo
> Sent: Friday, May 27, 2011 7:28 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Transferring to a cell phone
>
>
> Yungwei Chen wrote:
> >
> > Thanks for your reply. Using bridge fixed the problem. But I cannot hear
> > anything both ways. Any idea?
> After switching to using bridge function, does this also happen when you
> make an outbound call to your cell phone using your javascript?
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3
> Watts of electricity.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Transferring-to-a-cell-phone-tp6408246p6410887.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Fri Jun  3 20:57:05 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 3 Jun 2011 09:57:05 -0700
Subject: [Freeswitch-users] Freeswitch Server getting down
In-Reply-To: <BANLkTineWK6wstnKzdbM=rC31D9OBOfwZA@mail.gmail.com>
References: <BANLkTineWK6wstnKzdbM=rC31D9OBOfwZA@mail.gmail.com>
Message-ID: <BANLkTikc=Oh6tf4bMwX6nD481W_AX_a4JQ@mail.gmail.com>

I think Moises from Sangoma will want to check this out. First, though, be
sure that you do a "make current" and get on the latest code, then re-test.
Be prepared to collect logs, PRI traces, and/or backtrace info should the
devs ask for it.

-MC

On Thu, Jun 2, 2011 at 3:33 AM, ovvenkat <ovvenkatesan at gmail.com> wrote:

> Hi to all,
>
> I dont know whether this problem related to PRI line (OR) wanpipe driver (
> OR ) PRI Card
>
> I use Linux fedora 13  and Sangoma A101 PRI  card to handle incoming calls
> from E1 lines.
>
> Everything works fine for some time.
>
> After that, I could see  cli debug output saying,
>
> all the channels are disconnected and then immediately
> it says that "signaling status changed to UP".
>
> After some time, freeswitch getting shutdown saying
> ftmod_wanpipe.c:967 [s1c31][1:16] Failed to read from sangoma device: No
> buffer space available (-65)
>
>
> I pastebin debug logs for the same.
>
> http://pastebin.freeswitch.org/16432
>
> I am failed to identify the problem.
> Can you anyone guide me what is going wrong?
>
>
>
> Regards
> Venkatesan OV.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From freeswitch at simpot.com  Fri Jun  3 21:49:19 2011
From: freeswitch at simpot.com (Dmitry Saratsky)
Date: Fri, 3 Jun 2011 20:49:19 +0300
Subject: [Freeswitch-users] Codec enforcment from dialplan?
Message-ID: <000401cc2216$90f08b10$b2d1a130$@com>

Hi All,

 

I have some codec-prefs in my FS configured globally for all profiles (In my
case: "global_codec_prefs=G729,PCMU,PCMA,GSM"/>)

In addition, I need to provide for some specific incoming channels other
codec order (PCMU,PCMA) that destined to FS itself. I can identify those
channels by destination number of incoming call (fax service) in my
dialplan.

I can enforce needed codec from dialplan successfully for my outgoing
channels by using (absolute_codec_string='PCMU,PCMA'), but I'm failing to do
the same for incoming channels.

So now I'm enforcing all incoming calls from my provider to different from
global order (<param name="inbound-codec-prefs" value="PCMU,PCMA"/>),
instead of doing this for fax calls only.

 

Any ideas?

 

 

Relevant part of profile config:

 

<profile name="providers">

  <gateways>

    <X-PRE-PROCESS cmd="include" data="providers/*.xml"/>

  </gateways>

 

  <settings>

    <param name="debug" value="0"/>

    <param name="shutdown-on-fail" value="true"/>

    <param name="auth-calls" value="false"/>

    <param name="sip-trace" value="no"/>

    <param name="disable-register" value="true"/>

    <param name="rfc2833-pt" value="101"/>

    <param name="dialplan" value="XML"/>

    <param name="context" value="providers"/>

    <param name="apply-inbound-acl" value="013SONUS-IPS"/>

    <param name="hold-music" value="$${hold_music}"/>

    <param name="stun-enabled" value="false"/>

    <param name="stun-auto-disable" value="true"/>

    <param name="manage-presence" value="false"/>

    <param name="nonce-ttl" value="60"/>

    <param name="dtmf-type" value="rfc2833"/>

    <param name="dtmf-duration" value="2000"/>

 

    <param name="rtp-timer-name" value="soft"/>

    <param name="rtp-timeout-sec" value="50"/>

    <param name="rtp-hold-timeout-sec" value="300"/>

 

    <param name="sip-port" value="$${sip_port}"/>

 

    <param name="rtp-ip" value="$${rtp_ip}"/>

    <param name="sip-ip" value="$${ext_ip}"/>

 

<!--    <param name="codec-prefs" value="$${global_codec_prefs}"/>-->

    <param name="inbound-codec-prefs" value="PCMU,PCMA,G729"/>

    <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>

    <param name="inbound-late-negotiation" value="false"/>

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

    <param name="disable-transcoding" value="false"/>

 

  </settings>

</profile>

 

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From b_ball_henry at hotmail.com  Fri Jun  3 22:10:15 2011
From: b_ball_henry at hotmail.com (Henry Huang)
Date: Sat, 4 Jun 2011 02:10:15 +0800
Subject: [Freeswitch-users] mod_skypopen is awesome issue with demo-ivr
	conference
In-Reply-To: <BANLkTimu3GAitdDLvk=tcU+ZYUuXAjKCrw@mail.gmail.com>
References: <BANLkTik8hpzwjo-Susi6-wFDAzPiDZ-Jyw@mail.gmail.com>
	<BANLkTimu3GAitdDLvk=tcU+ZYUuXAjKCrw@mail.gmail.com>
Message-ID: <BANLkTimmW0ZcGW2CwRkH+iQy-tQXHbCstA@mail.gmail.com>

But the DTMF press worked very well while I was under the Demo IVR, even
when I press "1001" I was able to get connected to extension 1001. It didn't
work only after I was transferred to the conference selection.

Henry

On Sat, Jun 4, 2011 at 12:35 AM, Michael Collins <msc at freeswitch.org> wrote:

> The monkeys have stopped screaming for everyone. :( I have been looking for
> an alternative sound file for this but haven't found anything I really like.
>
> Not sure about the DTMF, but from what I understand there have been issues
> with sending DTMFs from the Skype client. I haven't tried it myself so I
> will defer to those who have.
>
> -MC
>
> On Fri, Jun 3, 2011 at 9:19 AM, Henry Huang <b_ball_henry at hotmail.com>wrote:
>
>> Hi:
>>
>> The new interactive installation for mod_skypopen is a piece of cake. Made
>> live much easier. I was so excited and tried it out right after the
>> installation was done. But I found a few issues calling the demo-ivr from
>> skyTpe client.
>>
>> 1. The screaming monkey does not scream anymore.
>> 2. The conference that connects to the freeswitch.org doesn't respond to
>> dtmf pressed on the skype client. Ever since the second client is connected
>> to the conference, the conference tell the client it's muted. So I tried
>> push 0 and any other key to see if I get to unmute. But no, non of the key
>> press works.
>>
>> If anyone have previous experiences on these and fixed them, please do
>> share your methods.
>>
>> Sound quality with the new virtual sound drive is very good.
>>
>> Thanks,
>>
>> Henry Huang
>> US: +1(818)6885508 | ??(Taiwan): +886 933847619
>> Contact Me [image: LinkedIn]<http://www.linkedin.com/profile/view?id=4590380> [image:
>> Facebook] <http://www.facebook.com/profile.php?id=595148342> [image:
>> Twitter] <http://twitter.com/unicsolution>
>> IM [image: Google Talk] red_rain_seven at gmail.com [image: Skype]unicsolution [image:
>> MSN] b_ball_henry at hotmail.com
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri Jun  3 22:32:58 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 3 Jun 2011 11:32:58 -0700
Subject: [Freeswitch-users] mod_skypopen is awesome issue with demo-ivr
	conference
In-Reply-To: <BANLkTimmW0ZcGW2CwRkH+iQy-tQXHbCstA@mail.gmail.com>
References: <BANLkTik8hpzwjo-Susi6-wFDAzPiDZ-Jyw@mail.gmail.com>
	<BANLkTimu3GAitdDLvk=tcU+ZYUuXAjKCrw@mail.gmail.com>
	<BANLkTimmW0ZcGW2CwRkH+iQy-tQXHbCstA@mail.gmail.com>
Message-ID: <BANLkTi=ULE8PKLOJRhhZkah71pZu_M0UqQ@mail.gmail.com>

Try transferring to a local conference on your machine and see what happens.
Watch the fs_cli and see if the DTMFs show up or not. From there we can see
what's up...

-MC

On Fri, Jun 3, 2011 at 11:10 AM, Henry Huang <b_ball_henry at hotmail.com>wrote:

> But the DTMF press worked very well while I was under the Demo IVR, even
> when I press "1001" I was able to get connected to extension 1001. It didn't
> work only after I was transferred to the conference selection.
>
> Henry
>
> On Sat, Jun 4, 2011 at 12:35 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> The monkeys have stopped screaming for everyone. :( I have been looking
>> for an alternative sound file for this but haven't found anything I really
>> like.
>>
>> Not sure about the DTMF, but from what I understand there have been issues
>> with sending DTMFs from the Skype client. I haven't tried it myself so I
>> will defer to those who have.
>>
>> -MC
>>
>> On Fri, Jun 3, 2011 at 9:19 AM, Henry Huang <b_ball_henry at hotmail.com>wrote:
>>
>>> Hi:
>>>
>>> The new interactive installation for mod_skypopen is a piece of cake.
>>> Made live much easier. I was so excited and tried it out right after the
>>> installation was done. But I found a few issues calling the demo-ivr from
>>> skyTpe client.
>>>
>>> 1. The screaming monkey does not scream anymore.
>>> 2. The conference that connects to the freeswitch.org doesn't respond to
>>> dtmf pressed on the skype client. Ever since the second client is connected
>>> to the conference, the conference tell the client it's muted. So I tried
>>> push 0 and any other key to see if I get to unmute. But no, non of the key
>>> press works.
>>>
>>> If anyone have previous experiences on these and fixed them, please do
>>> share your methods.
>>>
>>> Sound quality with the new virtual sound drive is very good.
>>>
>>> Thanks,
>>>
>>> Henry Huang
>>> US: +1(818)6885508 | ??(Taiwan): +886 933847619
>>> Contact Me [image: LinkedIn]<http://www.linkedin.com/profile/view?id=4590380> [image:
>>> Facebook] <http://www.facebook.com/profile.php?id=595148342> [image:
>>> Twitter] <http://twitter.com/unicsolution>
>>> IM [image: Google Talk] red_rain_seven at gmail.com [image: Skype]unicsolution [image:
>>> MSN] b_ball_henry at hotmail.com
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jan.berger at video24.no  Fri Jun  3 22:47:40 2011
From: jan.berger at video24.no (Jan Berger)
Date: Fri, 3 Jun 2011 20:47:40 +0200
Subject: [Freeswitch-users] Vestec ASR: 3 New Language Models
In-Reply-To: <4DE83D51.8020909@vestec.com>
References: <4DE83D51.8020909@vestec.com>
Message-ID: <FFD26DF7AC524FB6BD335210DA7D02B2@dell9400>

Kashif, 

Do you provide British English? 

Jan

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kashif
Kahn
Sent: 3. juni 2011 03:48
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Vestec ASR: 3 New Language Models

Dear All,

We have launched three new acoustic models for use with Vestec ASR: (1) 
Australian English, (2) Chinese Mandarin, and (3) Indian English.

Vestec offers the best deal around for enabling sophisticated speech 
recognition with command-and-control type IVR applications. Our ASR 
supports SRGS grammar format, can be integrated using MRCP (v1 & v2) 
interface,  offers high recognition accuracy, and costs a fraction of 
conventional speech engines. We offer the ASR in two versions according 
to vocabulary size: 500 items vocabulary and 2,500 items vocabulary, per 
recognition. A starter kit is available for $25 from Vestec webstore: 
http://www.vestec.com/products

Please do not hesitate to contact me with any questions. We are looking 
forward to being of service.

Regards,
-Kashif

-- 
Kashif Kahn
VP Business Development
Vestec Inc
Waterloo, ON Canada
phone: +1 519 885-7615


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



From steveayre at gmail.com  Fri Jun  3 23:09:07 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 3 Jun 2011 20:09:07 +0100
Subject: [Freeswitch-users] Codec enforcment from dialplan?
In-Reply-To: <000401cc2216$90f08b10$b2d1a130$@com>
References: <000401cc2216$90f08b10$b2d1a130$@com>
Message-ID: <BANLkTimVYfXVuBrLsKqwVbc4KeaoHSgD-Q@mail.gmail.com>

The default behaviour is to select the codec for an incoming call *before*
it hits the dialplan.

To work around that you should enable late-negotiation on the sofia profile
taking the incoming call. That'll delay selecting a codec until media starts
for a bridge. The codec will then match the bleg if possible, and otherwise
will pick the preferred codec from your lists. The alternative would be to
have multiple profiles with different codec preferences on each.

-Steve



On 3 June 2011 18:49, Dmitry Saratsky <freeswitch at simpot.com> wrote:

> Hi All,
>
>
>
> I have some codec-prefs in my FS configured globally for all profiles (In
> my case: "global_codec_prefs=G729,PCMU,PCMA,GSM"/>)
>
> In addition, I need to provide for some specific incoming channels other
> codec order (PCMU,PCMA) that destined to FS itself. I can identify those
> channels by destination number of incoming call (fax service) in my
> dialplan.
>
> I can enforce needed codec from dialplan successfully for my outgoing
> channels by using (absolute_codec_string='PCMU,PCMA'), but I?m failing to do
> the same for incoming channels?
>
> So now I?m enforcing all incoming calls from my provider to different from
> global order (<param name="inbound-codec-prefs" value="PCMU,PCMA"/>),
> instead of doing this for fax calls only?
>
>
>
> Any ideas?
>
>
>
>
>
> Relevant part of profile config:
>
>
>
> <profile name="providers">
>
>   <gateways>
>
>     <X-PRE-PROCESS cmd="include" data="providers/*.xml"/>
>
>   </gateways>
>
>
>
>   <settings>
>
>     <param name="debug" value="0"/>
>
>     <param name="shutdown-on-fail" value="true"/>
>
>     <param name="auth-calls" value="false"/>
>
>     <param name="sip-trace" value="no"/>
>
>     <param name="disable-register" value="true"/>
>
>     <param name="rfc2833-pt" value="101"/>
>
>     <param name="dialplan" value="XML"/>
>
>     <param name="context" value="providers"/>
>
>     <param name="apply-inbound-acl" value="013SONUS-IPS"/>
>
>     <param name="hold-music" value="$${hold_music}"/>
>
>     <param name="stun-enabled" value="false"/>
>
>     <param name="stun-auto-disable" value="true"/>
>
>     <param name="manage-presence" value="false"/>
>
>     <param name="nonce-ttl" value="60"/>
>
>     <param name="dtmf-type" value="rfc2833"/>
>
>     <param name="dtmf-duration" value="2000"/>
>
>
>
>     <param name="rtp-timer-name" value="soft"/>
>
>     <param name="rtp-timeout-sec" value="50"/>
>
>     <param name="rtp-hold-timeout-sec" value="300"/>
>
>
>
>     <param name="sip-port" value="$${sip_port}"/>
>
>
>
>     <param name="rtp-ip" value="$${rtp_ip}"/>
>
>     <param name="sip-ip" value="$${ext_ip}"/>
>
>
>
> <!--    <param name="codec-prefs" value="$${global_codec_prefs}"/>-->
>
>     <param name="inbound-codec-prefs" value="PCMU,PCMA,G729"/>
>
>     <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
>
>     <param name="inbound-late-negotiation" value="false"/>
>
>     <param name="inbound-codec-negotiation" value="greedy"/>
>
>     <param name="disable-transcoding" value="false"/>
>
>
>
>   </settings>
>
> </profile>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri Jun  3 23:18:17 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 3 Jun 2011 12:18:17 -0700
Subject: [Freeswitch-users] Codec enforcment from dialplan?
In-Reply-To: <000401cc2216$90f08b10$b2d1a130$@com>
References: <000401cc2216$90f08b10$b2d1a130$@com>
Message-ID: <BANLkTimgiHxRFJP_aeyRbQnHKtWR4X6-cQ@mail.gmail.com>

FYI, I wrote some nice, gentle words about codec negotiation on the wiki:

http://wiki.freeswitch.org/wiki/Codec_negotiation#Introduction

Start there - it will help you understand codec negotiation in general,
which will give you the foundation you need to do what you're trying to do.

-MC

On Fri, Jun 3, 2011 at 10:49 AM, Dmitry Saratsky <freeswitch at simpot.com>wrote:

> Hi All,
>
>
>
> I have some codec-prefs in my FS configured globally for all profiles (In
> my case: "global_codec_prefs=G729,PCMU,PCMA,GSM"/>)
>
> In addition, I need to provide for some specific incoming channels other
> codec order (PCMU,PCMA) that destined to FS itself. I can identify those
> channels by destination number of incoming call (fax service) in my
> dialplan.
>
> I can enforce needed codec from dialplan successfully for my outgoing
> channels by using (absolute_codec_string='PCMU,PCMA'), but I?m failing to do
> the same for incoming channels?
>
> So now I?m enforcing all incoming calls from my provider to different from
> global order (<param name="inbound-codec-prefs" value="PCMU,PCMA"/>),
> instead of doing this for fax calls only?
>
>
>
> Any ideas?
>
>
>
>
>
> Relevant part of profile config:
>
>
>
> <profile name="providers">
>
>   <gateways>
>
>     <X-PRE-PROCESS cmd="include" data="providers/*.xml"/>
>
>   </gateways>
>
>
>
>   <settings>
>
>     <param name="debug" value="0"/>
>
>     <param name="shutdown-on-fail" value="true"/>
>
>     <param name="auth-calls" value="false"/>
>
>     <param name="sip-trace" value="no"/>
>
>     <param name="disable-register" value="true"/>
>
>     <param name="rfc2833-pt" value="101"/>
>
>     <param name="dialplan" value="XML"/>
>
>     <param name="context" value="providers"/>
>
>     <param name="apply-inbound-acl" value="013SONUS-IPS"/>
>
>     <param name="hold-music" value="$${hold_music}"/>
>
>     <param name="stun-enabled" value="false"/>
>
>     <param name="stun-auto-disable" value="true"/>
>
>     <param name="manage-presence" value="false"/>
>
>     <param name="nonce-ttl" value="60"/>
>
>     <param name="dtmf-type" value="rfc2833"/>
>
>     <param name="dtmf-duration" value="2000"/>
>
>
>
>     <param name="rtp-timer-name" value="soft"/>
>
>     <param name="rtp-timeout-sec" value="50"/>
>
>     <param name="rtp-hold-timeout-sec" value="300"/>
>
>
>
>     <param name="sip-port" value="$${sip_port}"/>
>
>
>
>     <param name="rtp-ip" value="$${rtp_ip}"/>
>
>     <param name="sip-ip" value="$${ext_ip}"/>
>
>
>
> <!--    <param name="codec-prefs" value="$${global_codec_prefs}"/>-->
>
>     <param name="inbound-codec-prefs" value="PCMU,PCMA,G729"/>
>
>     <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
>
>     <param name="inbound-late-negotiation" value="false"/>
>
>     <param name="inbound-codec-negotiation" value="greedy"/>
>
>     <param name="disable-transcoding" value="false"/>
>
>
>
>   </settings>
>
> </profile>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From joaocarlosleme at gmail.com  Fri Jun  3 23:22:23 2011
From: joaocarlosleme at gmail.com (Joao Leme)
Date: Fri, 3 Jun 2011 12:22:23 -0700
Subject: [Freeswitch-users]  FIFO not working properly
Message-ID: <BANLkTikGaBggWh1sei0X3W49G6LWTEoX2w@mail.gmail.com>

FIFO is not ringing any extension while first call out is not hang up. Even
with five different agents to answer the queue only one (the first to
answer) get to take the calls, one at a time. Why? It used to work just fine
before. What may be causing this behavior?
Thanks,
John
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From msc at freeswitch.org  Fri Jun  3 23:38:12 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 3 Jun 2011 12:38:12 -0700
Subject: [Freeswitch-users] FIFO not working properly
In-Reply-To: <BANLkTikGaBggWh1sei0X3W49G6LWTEoX2w@mail.gmail.com>
References: <BANLkTikGaBggWh1sei0X3W49G6LWTEoX2w@mail.gmail.com>
Message-ID: <BANLkTikRge-40f5=j2qHo-_QKxpkb7eYPQ@mail.gmail.com>

Malfunction! Need input!

Collect up all the relevant info and put it on pastebin. See
http://wiki.freeswitch.org/wiki/Reporting_Bugs for hints on how to gather
data and drop it into pastebin.

-MC

On Fri, Jun 3, 2011 at 12:22 PM, Joao Leme <joaocarlosleme at gmail.com> wrote:

> FIFO is not ringing any extension while first call out is not hang up. Even
> with five different agents to answer the queue only one (the first to
> answer) get to take the calls, one at a time. Why? It used to work just fine
> before. What may be causing this behavior?
> Thanks,
> John
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From freeswitch at simpot.com  Fri Jun  3 23:41:45 2011
From: freeswitch at simpot.com (Dmitry Saratsky)
Date: Fri, 3 Jun 2011 22:41:45 +0300
Subject: [Freeswitch-users] Codec enforcment from dialplan?
In-Reply-To: <BANLkTimVYfXVuBrLsKqwVbc4KeaoHSgD-Q@mail.gmail.com>
References: <000401cc2216$90f08b10$b2d1a130$@com>
	<BANLkTimVYfXVuBrLsKqwVbc4KeaoHSgD-Q@mail.gmail.com>
Message-ID: <000901cc2226$455ee3d0$d01cab70$@com>

Hi Steven,

 

Thanks for your input, in that case you are right - I need to do
late-negotiation.

I have already tried to work with late-negotiation, but I was failed
probably because I still not understand enough about configuring this in
right way.

As far as I do understand, I should:

1.       Configure (<param name="inbound-late-negotiation" value="true"/>)
on my external profile. This is clear for me.

2.       Is there any limitation for ("inbound-codec-negotiation" value) in
that case?

3.       Configure (<action application="set" data="inherit_codec=true"/>)
in dialplan. This is not clear. Where in dialpan? On A leg (before
"transfer" DID) or on B leg (in general part before "bridge")? 

4.       Is this enough, or I should use also additional settings in
dialplan (Maybe: <action application="export"
data="codec_string=${ep_codec_string}"/>)? Where?

 

Thanks,

Dmitry.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven
Ayre
Sent: 03 Jun 2011 22:09
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Codec enforcment from dialplan?

 

The default behaviour is to select the codec for an incoming call *before*
it hits the dialplan.

To work around that you should enable late-negotiation on the sofia profile
taking the incoming call. That'll delay selecting a codec until media starts
for a bridge. The codec will then match the bleg if possible, and otherwise
will pick the preferred codec from your lists. The alternative would be to
have multiple profiles with different codec preferences on each.

-Steve




On 3 June 2011 18:49, Dmitry Saratsky <freeswitch at simpot.com> wrote:

Hi All,

 

I have some codec-prefs in my FS configured globally for all profiles (In my
case: "global_codec_prefs=G729,PCMU,PCMA,GSM"/>)

In addition, I need to provide for some specific incoming channels other
codec order (PCMU,PCMA) that destined to FS itself. I can identify those
channels by destination number of incoming call (fax service) in my
dialplan.

I can enforce needed codec from dialplan successfully for my outgoing
channels by using (absolute_codec_string='PCMU,PCMA'), but I'm failing to do
the same for incoming channels.

So now I'm enforcing all incoming calls from my provider to different from
global order (<param name="inbound-codec-prefs" value="PCMU,PCMA"/>),
instead of doing this for fax calls only.

 

Any ideas?

 

 

Relevant part of profile config:

 

<profile name="providers">

  <gateways>

    <X-PRE-PROCESS cmd="include" data="providers/*.xml"/>

  </gateways>

 

  <settings>

    <param name="debug" value="0"/>

    <param name="shutdown-on-fail" value="true"/>

    <param name="auth-calls" value="false"/>

    <param name="sip-trace" value="no"/>

    <param name="disable-register" value="true"/>

    <param name="rfc2833-pt" value="101"/>

    <param name="dialplan" value="XML"/>

    <param name="context" value="providers"/>

    <param name="apply-inbound-acl" value="013SONUS-IPS"/>

    <param name="hold-music" value="$${hold_music}"/>

    <param name="stun-enabled" value="false"/>

    <param name="stun-auto-disable" value="true"/>

    <param name="manage-presence" value="false"/>

    <param name="nonce-ttl" value="60"/>

    <param name="dtmf-type" value="rfc2833"/>

    <param name="dtmf-duration" value="2000"/>

 

    <param name="rtp-timer-name" value="soft"/>

    <param name="rtp-timeout-sec" value="50"/>

    <param name="rtp-hold-timeout-sec" value="300"/>

 

    <param name="sip-port" value="$${sip_port}"/>

 

    <param name="rtp-ip" value="$${rtp_ip}"/>

    <param name="sip-ip" value="$${ext_ip}"/>

 

<!--    <param name="codec-prefs" value="$${global_codec_prefs}"/>-->

    <param name="inbound-codec-prefs" value="PCMU,PCMA,G729"/>

    <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>

    <param name="inbound-late-negotiation" value="false"/>

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

    <param name="disable-transcoding" value="false"/>

 

  </settings>

</profile>

 


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

 

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From msc at freeswitch.org  Fri Jun  3 23:43:33 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 3 Jun 2011 12:43:33 -0700
Subject: [Freeswitch-users] Troubleshooting FreeSwitch Configuration
In-Reply-To: <BANLkTi=Qi2R5kXSUcy1Y4i8D++W5q4YzHA@mail.gmail.com>
References: <BANLkTi=Y5hGUGdOzxGQXOTzty-m0msutvA@mail.gmail.com>
	<BANLkTimZ4Ck7rfFeRRgxSu0-U_0OG19PcA@mail.gmail.com>
	<BANLkTi=Qi2R5kXSUcy1Y4i8D++W5q4YzHA@mail.gmail.com>
Message-ID: <BANLkTikatX752vhucKizjWA8WGEYfFdkLA@mail.gmail.com>

Interesting... I had never heard of "Karoo Bridge" - I'll have to contact
them and see if they will sponsor ClueCon. :P

If you don't have access to the fs_cli then just use tcpdump and grab the
port 5060 traffic (or whatever port your SIP traffic is on) and check for
clues. Once you know where the problem is then you can work on fixing it.

-MC

On Thu, Jun 2, 2011 at 5:33 PM, Roman Gelfand <rgelfand2 at gmail.com> wrote:

> Thanks for your help.
>
> The actual application I am using is Karoo Bridge SBC which is built
> on FreeSwitch.  I found freeswitch.xml.fsxml configuration file.  I
> suppose this the final resolved configuraiton.  I am not sure where I
> set debug option.
>
> Anyway, I thought the following could be useful for you to see.
>
>
> 09:03:38.274:   [CID=29f64488] B2B Transaction CREATED -
> register13166648z9hG4bKr84Dr2gK3ccHm
> 09:03:38.274:   [CID=29f64488] JS: Setting transaction property
> respond-to-packet-source=1
> 09:03:38.274:   [CID=29f64488] JS: Setting transaction property
> auth-method=access-list
> 09:03:38.274:   [CID=29f64488] JS: Setting transaction property
> auth-method=none
> 09:03:38.310:   [CID=29f64488] >>> REGISTER
> sip:callcentric.com;transport=udp SIP/2.0 LEN: 843 SRC:
> XX.XX.XXX.XXX:5060 DST: 204.11.192.37:5060 ENC: 0 PROT: udp
> 09:03:38.310: ^M
> {^M
>                [CID=29f64488] REGISTER
> sip:callcentric.com;transport=udp SIP/2.0^M
>                [CID=29f64488] From:
> <sip:1##########@callcentric.com;transport=udp>;tag=v0r8KQHmF054m^M
>                [CID=29f64488] To:
> <sip:1##########@callcentric.com;transport=udp>^M
>                [CID=29f64488] Call-ID:
> 99b2a5a7-1e7c-4e65-b9ac-6fadbf4ca1c3^M
>                [CID=29f64488] CSeq: 13166648 REGISTER^M
>                [CID=29f64488] Expires: 1800^M
>                [CID=29f64488] User-Agent: OSS Karoo FS^M
>                [CID=29f64488] Allow: INVITE, ACK, BYE, CANCEL,
> OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY^M
>                [CID=29f64488] Supported: timer, precondition, path,
> replaces^M
>                [CID=29f64488] Proxy-Authorization: Digest
> username="1##########", realm="callcentric.com",
> nonce="a0667bbc4a83b98f36b9619405daa1f5", opaque="", algorithm=MD5,
> uri="sip:XX.XX.XXX.XXX:5060",
> response="d4ee3820ca6f33ac8d40afc319b82006"^M
>                [CID=29f64488] Content-Length: 0^M
>                [CID=29f64488] Max-Forwards: 69^M
>                [CID=29f64488] Contact:
> <sip:1##########@XX.XX.XXX.XXX:5060;transport=udp>^M
>                [CID=29f64488] Via: SIP/2.0/UDP
>
> XX.XX.XXX.XXX:5060;branch=z9hG4bK34487872-8cf7-11e0-a33e-0800274986dc;rport^M
> };
> 09:03:38.279:   [CID=29f64488] B2B Transaction DESTROYED -
> register13166647z9hG4bKQZBNp7ZF63Nyr
> 09:03:38.348:   [CID=29f64488] Found Transaction
> register13166648z9hG4bK34487872-8cf7-11e0-a33e-0800274986dc
> 09:03:38.348:   [CID=29f64488] <<< SIP/2.0 407 Proxy Authentication
> Required LEN: 498 SRC: 204.11.192.37:5060 DST: XX.XX.XXX.XXX:5060 EXT:
> [XX.XX.XXX.XXX] ENC: 0 PROT: udp
> 09:03:38.348: ^M
> {^M
>                [CID=29f64488] SIP/2.0 407 Proxy Authentication Required^M
>                [CID=29f64488] v: SIP/2.0/UDP
>
> XX.XX.XXX.XXX:5060;branch=z9hG4bK34487872-8cf7-11e0-a33e-0800274986dc;rport=5060^M
>                [CID=29f64488] f:
> <sip:1##########@callcentric.com;transport=udp>;tag=v0r8KQHmF054m^M
>                [CID=29f64488] t:
> <sip:1##########@callcentric.com;transport=udp>^M
>                [CID=29f64488] i: 99b2a5a7-1e7c-4e65-b9ac-6fadbf4ca1c3^M
>                [CID=29f64488] CSeq: 13166648 REGISTER^M
>                [CID=29f64488] Proxy-Authenticate: Digest
> realm="callcentric.com", domain="sip:callcentric.com",
> nonce="edbfefbbc2ab6d684bae2b1de3fec2ea", opaque="", stale=TRUE,
> algorithm=MD5^M
>                [CID=29f64488] l: 0^M
> };
> 09:03:38.349: Missing contact header in REGISTER response.
> 09:03:38.350:   [CID=29f64488] >>> SIP/2.0 407 Proxy Authentication
> Required LEN: 526 SRC: XX.XX.XXX.XXX:5060 DST: XX.XX.XXX.XXX:5080 ENC:
> 0 PROT: udp
> 09:03:38.351: ^M
> {^M
>                [CID=29f64488] SIP/2.0 407 Proxy Authentication Required^M
>                [CID=29f64488] f:
> <sip:1##########@callcentric.com;transport=udp>;tag=v0r8KQHmF054m^M
>                [CID=29f64488] t:
> <sip:1##########@callcentric.com;transport=udp>^M
>                [CID=29f64488] i: 99b2a5a7-1e7c-4e65-b9ac-6fadbf4ca1c3^M
>                [CID=29f64488] CSeq: 13166648 REGISTER^M
>                [CID=29f64488] Proxy-Authenticate: Digest
> realm="callcentric.com", domain="sip:callcentric.com",
> nonce="edbfefbbc2ab6d684bae2b1de3fec2ea", opaque="", stale=TRUE,
> algorithm=MD5^M
>                [CID=29f64488] l: 0^M
>
>
>
>
>
> On Thu, Jun 2, 2011 at 5:57 PM, Steven Ayre <steveayre at gmail.com> wrote:
> > Sounds like they're not sending back a valid response.
> >
> > It would be useful to see what's beeing sent back and forth. Change the
> > logging level to debug and do 'sofia global siptrace on' to log all sip
> > packets so we can see what's in them. It could be a problem with their
> > server not sending the correct reply or it could be that your
> configuration
> > isn't quite right to interop with their server and needs tweaking.
> >
> > -Steve
> >
> >
> > On 2 June 2011 21:33, Roman Gelfand <rgelfand2 at gmail.com> wrote:
> >>
> >> I am running into trouble trying to register to ITSP.  The log file
> shows
> >> ...
> >>
> >> 2011-06-01 21:58:10.950068 [WARNING] sofia_reg.c:401 Timeout
> >> Registering 1XXXXXXXXXX-callcentric.com
> >> 2011-06-01 21:58:11.957417 [WARNING] sofia_reg.c:425
> >> 1XXXXXXXXXX-callcentric.com Failed Registration [0], setting retry to
> >> 30 seconds.
> >> 2011-06-01 21:58:42.046331 [NOTICE] sofia_reg.c:367 Registering
> >> 1XXXXXXXXXX-callcentric.com
> >>
> >> When challenged by 407 message from ITSP, I get
> >>
> >> 05:03:12.857: Missing contact header in REGISTER response.
> >>
> >> Does anyone know what is happening here? or pointers how to troubleshoot
> >> this?
> >>
> >> Thanks in advance
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From freeswitch at simpot.com  Fri Jun  3 23:44:29 2011
From: freeswitch at simpot.com (Dmitry Saratsky)
Date: Fri, 3 Jun 2011 22:44:29 +0300
Subject: [Freeswitch-users] Codec enforcment from dialplan?
In-Reply-To: <BANLkTimgiHxRFJP_aeyRbQnHKtWR4X6-cQ@mail.gmail.com>
References: <000401cc2216$90f08b10$b2d1a130$@com>
	<BANLkTimgiHxRFJP_aeyRbQnHKtWR4X6-cQ@mail.gmail.com>
Message-ID: <000e01cc2226$a70d7380$f5285a80$@com>

Hi Michael,

 

Thanks you, but I have already read this several time. Probably I still not
understand how do I configure late-negotiation.

Please see:
http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073246.html

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: 03 Jun 2011 22:18
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Codec enforcment from dialplan?

 

FYI, I wrote some nice, gentle words about codec negotiation on the wiki:

 

http://wiki.freeswitch.org/wiki/Codec_negotiation#Introduction

 

Start there - it will help you understand codec negotiation in general,
which will give you the foundation you need to do what you're trying to do.

 

-MC

On Fri, Jun 3, 2011 at 10:49 AM, Dmitry Saratsky <freeswitch at simpot.com>
wrote:

Hi All,

 

I have some codec-prefs in my FS configured globally for all profiles (In my
case: "global_codec_prefs=G729,PCMU,PCMA,GSM"/>)

In addition, I need to provide for some specific incoming channels other
codec order (PCMU,PCMA) that destined to FS itself. I can identify those
channels by destination number of incoming call (fax service) in my
dialplan.

I can enforce needed codec from dialplan successfully for my outgoing
channels by using (absolute_codec_string='PCMU,PCMA'), but I'm failing to do
the same for incoming channels.

So now I'm enforcing all incoming calls from my provider to different from
global order (<param name="inbound-codec-prefs" value="PCMU,PCMA"/>),
instead of doing this for fax calls only.

 

Any ideas?

 

 

Relevant part of profile config:

 

<profile name="providers">

  <gateways>

    <X-PRE-PROCESS cmd="include" data="providers/*.xml"/>

  </gateways>

 

  <settings>

    <param name="debug" value="0"/>

    <param name="shutdown-on-fail" value="true"/>

    <param name="auth-calls" value="false"/>

    <param name="sip-trace" value="no"/>

    <param name="disable-register" value="true"/>

    <param name="rfc2833-pt" value="101"/>

    <param name="dialplan" value="XML"/>

    <param name="context" value="providers"/>

    <param name="apply-inbound-acl" value="013SONUS-IPS"/>

    <param name="hold-music" value="$${hold_music}"/>

    <param name="stun-enabled" value="false"/>

    <param name="stun-auto-disable" value="true"/>

    <param name="manage-presence" value="false"/>

    <param name="nonce-ttl" value="60"/>

    <param name="dtmf-type" value="rfc2833"/>

    <param name="dtmf-duration" value="2000"/>

 

    <param name="rtp-timer-name" value="soft"/>

    <param name="rtp-timeout-sec" value="50"/>

    <param name="rtp-hold-timeout-sec" value="300"/>

 

    <param name="sip-port" value="$${sip_port}"/>

 

    <param name="rtp-ip" value="$${rtp_ip}"/>

    <param name="sip-ip" value="$${ext_ip}"/>

 

<!--    <param name="codec-prefs" value="$${global_codec_prefs}"/>-->

    <param name="inbound-codec-prefs" value="PCMU,PCMA,G729"/>

    <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>

    <param name="inbound-late-negotiation" value="false"/>

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

    <param name="disable-transcoding" value="false"/>

 

  </settings>

</profile>

 


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

 

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From msc at freeswitch.org  Fri Jun  3 23:48:56 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 3 Jun 2011 12:48:56 -0700
Subject: [Freeswitch-users] Codec enforcment from dialplan?
In-Reply-To: <000901cc2226$455ee3d0$d01cab70$@com>
References: <000401cc2216$90f08b10$b2d1a130$@com>
	<BANLkTimVYfXVuBrLsKqwVbc4KeaoHSgD-Q@mail.gmail.com>
	<000901cc2226$455ee3d0$d01cab70$@com>
Message-ID: <BANLkTik1bvne6x8tq8wCXgkwdQgVRFJw2w@mail.gmail.com>

You may have replied here before you saw my last email, but definitely go
read the codec negotiation page on the wiki. There you will find out whether
or not you need "inherit_codec" or something else. It all depends on exactly
what you are trying to accomplish.

-MC

On Fri, Jun 3, 2011 at 12:41 PM, Dmitry Saratsky <freeswitch at simpot.com>wrote:

> Hi Steven,
>
>
>
> Thanks for your input, in that case you are right ? I need to do
> late-negotiation.
>
> I have already tried to work with late-negotiation, but I was failed
> probably because I still not understand enough about configuring this in
> right way?
>
> As far as I do understand, I should:
>
> 1.       Configure (<param name="inbound-late-negotiation" value="true"/>)
> on my external profile. This is clear for me.
>
> 2.       Is there any limitation for ("inbound-codec-negotiation" value)
> in that case?
>
> 3.       Configure (<action application="set" data="inherit_codec=true"/>)
> in dialplan. This is not clear? Where in dialpan? On A leg (before
> ?transfer? DID) or on B leg (in general part before ?bridge?)?
>
> 4.       Is this enough, or I should use also additional settings in
> dialplan (Maybe: <action application="export"
> data="codec_string=${ep_codec_string}"/>)? Where?
>
>
>
> Thanks,
>
> Dmitry.
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* 03 Jun 2011 22:09
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Codec enforcment from dialplan?
>
>
>
> The default behaviour is to select the codec for an incoming call *before*
> it hits the dialplan.
>
> To work around that you should enable late-negotiation on the sofia profile
> taking the incoming call. That'll delay selecting a codec until media starts
> for a bridge. The codec will then match the bleg if possible, and otherwise
> will pick the preferred codec from your lists. The alternative would be to
> have multiple profiles with different codec preferences on each.
>
> -Steve
>
>
> On 3 June 2011 18:49, Dmitry Saratsky <freeswitch at simpot.com> wrote:
>
> Hi All,
>
>
>
> I have some codec-prefs in my FS configured globally for all profiles (In
> my case: "global_codec_prefs=G729,PCMU,PCMA,GSM"/>)
>
> In addition, I need to provide for some specific incoming channels other
> codec order (PCMU,PCMA) that destined to FS itself. I can identify those
> channels by destination number of incoming call (fax service) in my
> dialplan.
>
> I can enforce needed codec from dialplan successfully for my outgoing
> channels by using (absolute_codec_string='PCMU,PCMA'), but I?m failing to do
> the same for incoming channels?
>
> So now I?m enforcing all incoming calls from my provider to different from
> global order (<param name="inbound-codec-prefs" value="PCMU,PCMA"/>),
> instead of doing this for fax calls only?
>
>
>
> Any ideas?
>
>
>
>
>
> Relevant part of profile config:
>
>
>
> <profile name="providers">
>
>   <gateways>
>
>     <X-PRE-PROCESS cmd="include" data="providers/*.xml"/>
>
>   </gateways>
>
>
>
>   <settings>
>
>     <param name="debug" value="0"/>
>
>     <param name="shutdown-on-fail" value="true"/>
>
>     <param name="auth-calls" value="false"/>
>
>     <param name="sip-trace" value="no"/>
>
>     <param name="disable-register" value="true"/>
>
>     <param name="rfc2833-pt" value="101"/>
>
>     <param name="dialplan" value="XML"/>
>
>     <param name="context" value="providers"/>
>
>     <param name="apply-inbound-acl" value="013SONUS-IPS"/>
>
>     <param name="hold-music" value="$${hold_music}"/>
>
>     <param name="stun-enabled" value="false"/>
>
>     <param name="stun-auto-disable" value="true"/>
>
>     <param name="manage-presence" value="false"/>
>
>     <param name="nonce-ttl" value="60"/>
>
>     <param name="dtmf-type" value="rfc2833"/>
>
>     <param name="dtmf-duration" value="2000"/>
>
>
>
>     <param name="rtp-timer-name" value="soft"/>
>
>     <param name="rtp-timeout-sec" value="50"/>
>
>     <param name="rtp-hold-timeout-sec" value="300"/>
>
>
>
>     <param name="sip-port" value="$${sip_port}"/>
>
>
>
>     <param name="rtp-ip" value="$${rtp_ip}"/>
>
>     <param name="sip-ip" value="$${ext_ip}"/>
>
>
>
> <!--    <param name="codec-prefs" value="$${global_codec_prefs}"/>-->
>
>     <param name="inbound-codec-prefs" value="PCMU,PCMA,G729"/>
>
>     <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
>
>     <param name="inbound-late-negotiation" value="false"/>
>
>     <param name="inbound-codec-negotiation" value="greedy"/>
>
>     <param name="disable-transcoding" value="false"/>
>
>
>
>   </settings>
>
> </profile>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri Jun  3 23:53:29 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 3 Jun 2011 12:53:29 -0700
Subject: [Freeswitch-users] FIFO suddenly stopped working properly
In-Reply-To: <BANLkTim8MJUDxc+0aQusxckV4+X6mrHwDA@mail.gmail.com>
References: <BANLkTim8MJUDxc+0aQusxckV4+X6mrHwDA@mail.gmail.com>
Message-ID: <BANLkTin72-yCR=gzG1+trJ3i63h_kzsn_A@mail.gmail.com>

Also, check out this thread, particularly the comments from Anthony:
http://lists.freeswitch.org/pipermail/freeswitch-users/2011-March/070588.html

If you had a really old version of FIFO and updated then you may have
crossed over from when caller ID was not the default to the current state
where it is the default.

-MC

On Thu, Jun 2, 2011 at 2:02 PM, Joao Leme <joaocarlosleme at gmail.com> wrote:

> Hi there,
> No change nor updates have been made in months. Everything was working
> great but now FIFO is not ringing the extensions while the first call is not
> HUNG UP...meaning, first call to go to FIFO rings the agents, and one
> person answer...when the second call comes in, it goes to fifo queue (caller
> listens the music) but no one is aware of the call, nor even the agents that
> are available. Only after the first call hangs up the 2nd goes in. I tried
> troubleshooting but no help.
>
> What I also noticed is that before, to display the caller ID on my sip
> client I had to use:
>
> <action application="set"
> data="origination_caller_id_number=${caller_id_number}"/>
> <action application="set"
> data="origination_caller_id_name=${caller_id_name}"/>
>
> but now, while testing, the caller id was showing when calling
> fifo without setting those parameters, but before it would just show the
> fifo queue name.
>
> All I can say is that I can't figure out why and how to fix these behavior
> if I have done no change to FreeSWITCH nor the config files. I also tried
> downloading the latest GIT and got the same response.
>
> THANKS,
> John
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gmaruzz at gmail.com  Sat Jun  4 01:08:51 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 3 Jun 2011 23:08:51 +0200
Subject: [Freeswitch-users] mod_skypopen is awesome issue with demo-ivr
	conference
In-Reply-To: <BANLkTi=ULE8PKLOJRhhZkah71pZu_M0UqQ@mail.gmail.com>
References: <BANLkTik8hpzwjo-Susi6-wFDAzPiDZ-Jyw@mail.gmail.com>
	<BANLkTimu3GAitdDLvk=tcU+ZYUuXAjKCrw@mail.gmail.com>
	<BANLkTimmW0ZcGW2CwRkH+iQy-tQXHbCstA@mail.gmail.com>
	<BANLkTi=ULE8PKLOJRhhZkah71pZu_M0UqQ@mail.gmail.com>
Message-ID: <BANLkTinOpAXeV3wWMS=dHekQ-rw_d9urxA@mail.gmail.com>

As per the wiki page, and as is clearly written in the console and the
logfile, if an incomong skype call is bridged to an outbound call, the
dtmf is by default passed only in band (eg: as audio).
If you want to have it working with the fs developer conference,
create an extension that goes there and add to it the variable that's
in the wiki page (can't remember the name, is something like
"dtmf-outband-also-when-bridged=true").
Please check the wiki page for the variable name.
Reason for this is that the in band dtmf cannot be shut up, and often
is detected by the remote party. But in this particular case of the fs
developer conference, the remote party is setup to not detect inband
in sip (or anyway, it does not detect it).
Long story short: create an extension that sets that variable and goes
to the fs conference and you'll be all set.
Tomorrow I'll post here an example from my dialplan.
-giovanni

On 6/3/11, Michael Collins <msc at freeswitch.org> wrote:
> Try transferring to a local conference on your machine and see what happens.
> Watch the fs_cli and see if the DTMFs show up or not. From there we can see
> what's up...
>
> -MC
>
> On Fri, Jun 3, 2011 at 11:10 AM, Henry Huang
> <b_ball_henry at hotmail.com>wrote:
>
>> But the DTMF press worked very well while I was under the Demo IVR, even
>> when I press "1001" I was able to get connected to extension 1001. It
>> didn't
>> work only after I was transferred to the conference selection.
>>
>> Henry
>>
>> On Sat, Jun 4, 2011 at 12:35 AM, Michael Collins
>> <msc at freeswitch.org>wrote:
>>
>>> The monkeys have stopped screaming for everyone. :( I have been looking
>>> for an alternative sound file for this but haven't found anything I
>>> really
>>> like.
>>>
>>> Not sure about the DTMF, but from what I understand there have been
>>> issues
>>> with sending DTMFs from the Skype client. I haven't tried it myself so I
>>> will defer to those who have.
>>>
>>> -MC
>>>
>>> On Fri, Jun 3, 2011 at 9:19 AM, Henry Huang
>>> <b_ball_henry at hotmail.com>wrote:
>>>
>>>> Hi:
>>>>
>>>> The new interactive installation for mod_skypopen is a piece of cake.
>>>> Made live much easier. I was so excited and tried it out right after the
>>>> installation was done. But I found a few issues calling the demo-ivr
>>>> from
>>>> skyTpe client.
>>>>
>>>> 1. The screaming monkey does not scream anymore.
>>>> 2. The conference that connects to the freeswitch.org doesn't respond to
>>>> dtmf pressed on the skype client. Ever since the second client is
>>>> connected
>>>> to the conference, the conference tell the client it's muted. So I tried
>>>> push 0 and any other key to see if I get to unmute. But no, non of the
>>>> key
>>>> press works.
>>>>
>>>> If anyone have previous experiences on these and fixed them, please do
>>>> share your methods.
>>>>
>>>> Sound quality with the new virtual sound drive is very good.
>>>>
>>>> Thanks,
>>>>
>>>> Henry Huang
>>>> US: +1(818)6885508 | ??(Taiwan): +886 933847619
>>>> Contact Me [image:
>>>> LinkedIn]<http://www.linkedin.com/profile/view?id=4590380> [image:
>>>> Facebook] <http://www.facebook.com/profile.php?id=595148342> [image:
>>>> Twitter] <http://twitter.com/unicsolution>
>>>> IM [image: Google Talk] red_rain_seven at gmail.com [image:
>>>> Skype]unicsolution [image:
>>>> MSN] b_ball_henry at hotmail.com
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From rgelfand2 at gmail.com  Sat Jun  4 01:42:27 2011
From: rgelfand2 at gmail.com (Roman Gelfand)
Date: Fri, 3 Jun 2011 17:42:27 -0400
Subject: [Freeswitch-users] Troubleshooting FreeSwitch Configuration
In-Reply-To: <BANLkTikatX752vhucKizjWA8WGEYfFdkLA@mail.gmail.com>
References: <BANLkTi=Y5hGUGdOzxGQXOTzty-m0msutvA@mail.gmail.com>
	<BANLkTimZ4Ck7rfFeRRgxSu0-U_0OG19PcA@mail.gmail.com>
	<BANLkTi=Qi2R5kXSUcy1Y4i8D++W5q4YzHA@mail.gmail.com>
	<BANLkTikatX752vhucKizjWA8WGEYfFdkLA@mail.gmail.com>
Message-ID: <BANLkTi=_U1JmQfs3oA=2crBgNDfKR6ccuA@mail.gmail.com>

As it turns out, I do have access to fs_cli.  In fact when I left it,
it gave me this message

[INFO] libs/esl/fs_cli.c:676 process_command() Goodbye!
See you at ClueCon http://www.cluecon.com/

I am looking into it right now.  Thanks for the help.

On Fri, Jun 3, 2011 at 3:43 PM, Michael Collins <msc at freeswitch.org> wrote:
> Interesting... I had never heard of "Karoo Bridge" - I'll have to contact
> them and see if they will sponsor ClueCon. :P
> If you don't have access to the fs_cli then just use tcpdump and grab the
> port 5060 traffic (or whatever port your SIP traffic is on) and check for
> clues. Once you know where the problem is then you can work on fixing it.
> -MC
> On Thu, Jun 2, 2011 at 5:33 PM, Roman Gelfand <rgelfand2 at gmail.com> wrote:
>>
>> Thanks for your help.
>>
>> The actual application I am using is Karoo Bridge SBC which is built
>> on FreeSwitch. ?I found freeswitch.xml.fsxml configuration file. ?I
>> suppose this the final resolved configuraiton. ?I am not sure where I
>> set debug option.
>>
>> Anyway, I thought the following could be useful for you to see.
>>
>>
>> 09:03:38.274: ? [CID=29f64488] B2B Transaction CREATED -
>> register13166648z9hG4bKr84Dr2gK3ccHm
>> 09:03:38.274: ? [CID=29f64488] JS: Setting transaction property
>> respond-to-packet-source=1
>> 09:03:38.274: ? [CID=29f64488] JS: Setting transaction property
>> auth-method=access-list
>> 09:03:38.274: ? [CID=29f64488] JS: Setting transaction property
>> auth-method=none
>> 09:03:38.310: ? [CID=29f64488] >>> REGISTER
>> sip:callcentric.com;transport=udp SIP/2.0 LEN: 843 SRC:
>> XX.XX.XXX.XXX:5060 DST: 204.11.192.37:5060 ENC: 0 PROT: udp
>> 09:03:38.310: ^M
>> {^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] REGISTER
>> sip:callcentric.com;transport=udp SIP/2.0^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] From:
>> <sip:1##########@callcentric.com;transport=udp>;tag=v0r8KQHmF054m^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] To:
>> <sip:1##########@callcentric.com;transport=udp>^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] Call-ID:
>> 99b2a5a7-1e7c-4e65-b9ac-6fadbf4ca1c3^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] CSeq: 13166648 REGISTER^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] Expires: 1800^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] User-Agent: OSS Karoo FS^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] Allow: INVITE, ACK, BYE, CANCEL,
>> OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] Supported: timer, precondition, path,
>> replaces^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] Proxy-Authorization: Digest
>> username="1##########", realm="callcentric.com",
>> nonce="a0667bbc4a83b98f36b9619405daa1f5", opaque="", algorithm=MD5,
>> uri="sip:XX.XX.XXX.XXX:5060",
>> response="d4ee3820ca6f33ac8d40afc319b82006"^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] Content-Length: 0^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] Max-Forwards: 69^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] Contact:
>> <sip:1##########@XX.XX.XXX.XXX:5060;transport=udp>^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] Via: SIP/2.0/UDP
>>
>> XX.XX.XXX.XXX:5060;branch=z9hG4bK34487872-8cf7-11e0-a33e-0800274986dc;rport^M
>> };
>> 09:03:38.279: ? [CID=29f64488] B2B Transaction DESTROYED -
>> register13166647z9hG4bKQZBNp7ZF63Nyr
>> 09:03:38.348: ? [CID=29f64488] Found Transaction
>> register13166648z9hG4bK34487872-8cf7-11e0-a33e-0800274986dc
>> 09:03:38.348: ? [CID=29f64488] <<< SIP/2.0 407 Proxy Authentication
>> Required LEN: 498 SRC: 204.11.192.37:5060 DST: XX.XX.XXX.XXX:5060 EXT:
>> [XX.XX.XXX.XXX] ENC: 0 PROT: udp
>> 09:03:38.348: ^M
>> {^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] SIP/2.0 407 Proxy Authentication Required^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] v: SIP/2.0/UDP
>>
>> XX.XX.XXX.XXX:5060;branch=z9hG4bK34487872-8cf7-11e0-a33e-0800274986dc;rport=5060^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] f:
>> <sip:1##########@callcentric.com;transport=udp>;tag=v0r8KQHmF054m^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] t:
>> <sip:1##########@callcentric.com;transport=udp>^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] i: 99b2a5a7-1e7c-4e65-b9ac-6fadbf4ca1c3^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] CSeq: 13166648 REGISTER^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] Proxy-Authenticate: Digest
>> realm="callcentric.com", domain="sip:callcentric.com",
>> nonce="edbfefbbc2ab6d684bae2b1de3fec2ea", opaque="", stale=TRUE,
>> algorithm=MD5^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] l: 0^M
>> };
>> 09:03:38.349: Missing contact header in REGISTER response.
>> 09:03:38.350: ? [CID=29f64488] >>> SIP/2.0 407 Proxy Authentication
>> Required LEN: 526 SRC: XX.XX.XXX.XXX:5060 DST: XX.XX.XXX.XXX:5080 ENC:
>> 0 PROT: udp
>> 09:03:38.351: ^M
>> {^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] SIP/2.0 407 Proxy Authentication Required^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] f:
>> <sip:1##########@callcentric.com;transport=udp>;tag=v0r8KQHmF054m^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] t:
>> <sip:1##########@callcentric.com;transport=udp>^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] i: 99b2a5a7-1e7c-4e65-b9ac-6fadbf4ca1c3^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] CSeq: 13166648 REGISTER^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] Proxy-Authenticate: Digest
>> realm="callcentric.com", domain="sip:callcentric.com",
>> nonce="edbfefbbc2ab6d684bae2b1de3fec2ea", opaque="", stale=TRUE,
>> algorithm=MD5^M
>> ? ? ? ? ? ? ? ?[CID=29f64488] l: 0^M
>>
>>
>>
>>
>>
>> On Thu, Jun 2, 2011 at 5:57 PM, Steven Ayre <steveayre at gmail.com> wrote:
>> > Sounds like they're not sending back a valid response.
>> >
>> > It would be useful to see what's beeing sent back and forth. Change the
>> > logging level to debug and do 'sofia global siptrace on' to log all sip
>> > packets so we can see what's in them. It could be a problem with their
>> > server not sending the correct reply or it could be that your
>> > configuration
>> > isn't quite right to interop with their server and needs tweaking.
>> >
>> > -Steve
>> >
>> >
>> > On 2 June 2011 21:33, Roman Gelfand <rgelfand2 at gmail.com> wrote:
>> >>
>> >> I am running into trouble trying to register to ITSP. ?The log file
>> >> shows
>> >> ...
>> >>
>> >> 2011-06-01 21:58:10.950068 [WARNING] sofia_reg.c:401 Timeout
>> >> Registering 1XXXXXXXXXX-callcentric.com
>> >> 2011-06-01 21:58:11.957417 [WARNING] sofia_reg.c:425
>> >> 1XXXXXXXXXX-callcentric.com Failed Registration [0], setting retry to
>> >> 30 seconds.
>> >> 2011-06-01 21:58:42.046331 [NOTICE] sofia_reg.c:367 Registering
>> >> 1XXXXXXXXXX-callcentric.com
>> >>
>> >> When challenged by 407 message from ITSP, I get
>> >>
>> >> 05:03:12.857: Missing contact header in REGISTER response.
>> >>
>> >> Does anyone know what is happening here? or pointers how to
>> >> troubleshoot
>> >> this?
>> >>
>> >> Thanks in advance
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From rgelfand2 at gmail.com  Sat Jun  4 02:04:47 2011
From: rgelfand2 at gmail.com (Roman Gelfand)
Date: Fri, 3 Jun 2011 18:04:47 -0400
Subject: [Freeswitch-users] Need help interpreting fs_cli debug messages
Message-ID: <BANLkTingE3hspOajHyB12aszr68rgit_KA@mail.gmail.com>

I was wondering if somebody could help interpret the registration issue.

+OK log level  [7]
freeswitch at internal> nua: nh_create_handle: entering
nua: nua_handle_bind: entering
2011-06-02 13:35:18.401497 [NOTICE] sofia_reg.c:367 Registering
###########-callcentric.com
nua: nua_register: entering
nua(0x873dc88): sent signal r_register
nua: nua_stack_set_params: entering
soa_clone(static::0x86d2e30, 0x86fba08, 0x873dc88) called
soa_set_params(static::0x8dd11008, ...) called
soa_set_params(static::0x8dd11008, ...) called
nua(0x873dc88): adding register usage
nta_leg_tcreate(0x8dd16398)
nta: selecting scheme sip
tport_tsend(0x86afae0) tpn = */XX.XX.XXX.XXX:5060
tport_resolve addrinfo = XX.XX.XXX.XXX:5060
tport_by_addrinfo(0x86afae0): not found by name */XX.XX.XXX.XXX:5060
tport_vsend returned 639
nta: sent REGISTER (13181995) to */XX.XX.XXX.XXX:5060
tport_pend(0x86afae0): pending 0x8dd0a588 for udp/XX.XX.XXX.XXX:5080 (already 0)
nta: timer set to 32000 ms
nta: timer shortened to 1000 ms
tport_wakeup_pri(0x86afae0): events IN
tport_recv_event(0x86afae0)
tport_recv_iovec(0x86afae0) msg 0x8dd08cf8 from
(udp/XX.XX.XXX.XXX:5080) has 526 bytes, veclen = 1
tport_deliver(0x86afae0): msg 0x8dd08cf8 (526 bytes) from
udp/XX.XX.XXX.XXX:5080/sip next=(nil)
nta: received 407 Proxy Authentication Required for REGISTER (13181995)
nta: 407 Proxy Authentication Required is going to a transaction
nta_outgoing: RTT is 135.683 ms
tport_release(0x86afae0): 0x8dd0a588 by 0x8dd009d8 with 0x8dd08cf8
auth_digest_challenge_get(): got 7
nta: outgoing_free(0x8dd009d8)
nua: nua_application_event: entering
nua: nua_authenticate: entering
nua(0x873dc88): sent signal r_authenticate
nua: nua_handle_magic: entering
auth_digest_a1() has A1 = MD5(###########:callcentric.com:passwd) =
8dddd49339918097d09e22dd7b15c38b
A2 = MD5(REGISTER:sip:XX.XX.XXX.XXX:5060)
auth_response: 9f45cc31d0328fdf18a7e4d3b547e3ea =
MD5(8dddd49339918097d09e22dd7b15c38b:dbcc6c9095c35c4be730ec1110f970e5:db5f9cddc4d0d297aa85cc660f20595a)
(qop=NONE)
nta: selecting scheme sip
tport_tsend(0x86afae0) tpn = */XX.XX.XXX.XXX:5060
tport_resolve addrinfo = XX.XX.XXX.XXX:5060
tport_by_addrinfo(0x86afae0): not found by name */XX.XX.XXX.XXX:5060
tport_vsend returned 859
nta: sent REGISTER (13181996) to */XX.XX.XXX.XXX:5060
tport_pend(0x86afae0): pending 0x8dd08cf8 for udp/XX.XX.XXX.XXX:5080 (already 0)
tport_wakeup_pri(0x86afae0): events IN
tport_recv_event(0x86afae0)
tport_recv_iovec(0x86afae0) msg 0x8dd1ae88 from
(udp/XX.XX.XXX.XXX:5080) has 526 bytes, veclen = 1
tport_deliver(0x86afae0): msg 0x8dd1ae88 (526 bytes) from
udp/XX.XX.XXX.XXX:5080/sip next=(nil)
nta: received 407 Proxy Authentication Required for REGISTER (13181996)
nta: 407 Proxy Authentication Required is going to a transaction
nta_outgoing: RTT is 127.055 ms
tport_release(0x86afae0): 0x8dd08cf8 by 0x8dd08a98 with 0x8dd1ae88
auth_digest_challenge_get(): got 7
auth_digest_a1() has A1 = MD5(###########:callcentric.com:passwd) =
8dddd49339918097d09e22dd7b15c38b
A2 = MD5(REGISTER:sip:XX.XX.XXX.XXX:5060)
auth_response: c0b3965ec27e2e3386277645ae14f324 =
MD5(8dddd49339918097d09e22dd7b15c38b:c76385015930949803ca8b0946f81d0e:db5f9cddc4d0d297aa85cc660f20595a)
(qop=NONE)
nta: selecting scheme sip
tport_tsend(0x86afae0) tpn = */XX.XX.XXX.XXX:5060
tport_resolve addrinfo = XX.XX.XXX.XXX:5060
tport_by_addrinfo(0x86afae0): not found by name */XX.XX.XXX.XXX:5060
tport_vsend returned 859
nta: sent REGISTER (13181997) to */XX.XX.XXX.XXX:5060
tport_pend(0x86afae0): pending 0x8dd19ea8 for udp/XX.XX.XXX.XXX:5080 (already 0)
nta: outgoing_free(0x8dd08a98)
nua: nua_application_event: entering
nua: nua_handle_magic: entering
tport_wakeup_pri(0x86afae0): events IN
tport_recv_event(0x86afae0)
tport_recv_iovec(0x86afae0) msg 0x8dd1ae88 from
(udp/XX.XX.XXX.XXX:5080) has 526 bytes, veclen = 1
tport_deliver(0x86afae0): msg 0x8dd1ae88 (526 bytes) from
udp/XX.XX.XXX.XXX:5080/sip next=(nil)
nta: received 407 Proxy Authentication Required for REGISTER (13181997)
nta: 407 Proxy Authentication Required is going to a transaction
nta_outgoing: RTT is 139.175 ms
tport_release(0x86afae0): 0x8dd19ea8 by 0x8dd18790 with 0x8dd1ae88
auth_digest_challenge_get(): got 7
auth_digest_a1() has A1 = MD5(###########:callcentric.com:passwd) =
8dddd49339918097d09e22dd7b15c38b
A2 = MD5(REGISTER:sip:XX.XX.XXX.XXX:5060)
auth_response: f014cc64fa858ec3615b2b1caab196a6 =
MD5(8dddd49339918097d09e22dd7b15c38b:274bba14cbca0384c30f227da6c42c35:db5f9cddc4d0d297aa85cc660f20595a)
(qop=NONE)
nta: selecting scheme sip
tport_tsend(0x86afae0) tpn = */XX.XX.XXX.XXX:5060
tport_resolve addrinfo = XX.XX.XXX.XXX:5060
tport_by_addrinfo(0x86afae0): not found by name */XX.XX.XXX.XXX:5060
tport_vsend returned 859
nta: sent REGISTER (13181998) to */XX.XX.XXX.XXX:5060
tport_pend(0x86afae0): pending 0x8dd18f80 for udp/XX.XX.XXX.XXX:5080 (already 0)
nta: outgoing_free(0x8dd18790)
nua: nua_application_event: entering
nua: nua_handle_magic: entering
tport_wakeup_pri(0x86afae0): events IN
tport_recv_event(0x86afae0)
tport_recv_iovec(0x86afae0) msg 0x8dd1ae88 from
(udp/XX.XX.XXX.XXX:5080) has 526 bytes, veclen = 1
tport_deliver(0x86afae0): msg 0x8dd1ae88 (526 bytes) from
udp/XX.XX.XXX.XXX:5080/sip next=(nil)
nta: received 407 Proxy Authentication Required for REGISTER (13181998)
nta: 407 Proxy Authentication Required is going to a transaction
nta_outgoing: RTT is 114.044 ms
tport_release(0x86afae0): 0x8dd18f80 by 0x8dd16e10 with 0x8dd1ae88
nta: outgoing_free(0x8dd16e10)
nua(0x873dc88): removing register usage
nta_leg_destroy(0x8dd16398)
nua: nua_application_event: entering
nua: nua_authenticate: entering
nua(0x873dc88): sent signal r_authenticate
nua: nua_handle_magic: entering
nua: nua_application_event: entering
nua: nua_handle_destroy: entering
nua(0x873dc88): sent signal r_destroy
nua: nua_handle_magic: entering
nua: nua_handle_bind: entering
nua: nua_handle_destroy: entering
nta_leg_destroy((nil))
soa_destroy(static::0x8dd11008) called
nta: timer not set
2011-06-02 13:36:19.875422 [WARNING] sofia_reg.c:401 Timeout
Registering ###########-callcentric.com
nua: nua_handle_destroy: entering
2011-06-02 13:36:20.881513 [WARNING] sofia_reg.c:425
###########-callcentric.com Failed Registration [0], setting retry to
30 seconds.


Thanks in advance


From joaocarlosleme at gmail.com  Sat Jun  4 02:31:27 2011
From: joaocarlosleme at gmail.com (Joao Leme)
Date: Fri, 3 Jun 2011 15:31:27 -0700
Subject: [Freeswitch-users] FIFO suddenly stopped working properly
In-Reply-To: <BANLkTin72-yCR=gzG1+trJ3i63h_kzsn_A@mail.gmail.com>
References: <BANLkTim8MJUDxc+0aQusxckV4+X6mrHwDA@mail.gmail.com>
	<BANLkTin72-yCR=gzG1+trJ3i63h_kzsn_A@mail.gmail.com>
Message-ID: <BANLkTik2QK=SH1Yjg1DPBVKbr7HNqLdD8g@mail.gmail.com>

Thanks Michael, but I don't know what I have to do to get it fixed. Is there
a configuration change? I'm not good at debugging and probably I should
change the subject to "How to configure FIFO properly" (hoping to be a
configuration problem).

Below is my simple configuration for FIFO:

fifo.conf.xml:
<fifo name="sales_fifo@$${domain}" importance="0">
      <member timeout="10" simo="10"
lag="3">{fifo_member_wait=nowait}group/attendants@$${domain}</member>
      <member timeout="300" simo="10"
lag="3">{fifo_member_wait=nowait}group/sales@$${domain}</member>
</fifo>

directory/default.xml:
<groups>
 <group name="sales">
        <users>
          <user id="1000" type="pointer"/>
          <user id="1001" type="pointer"/>
          <user id="1002" type="pointer"/>
          <user id="1003" type="pointer"/>
        </users>
      </group>
 <group name="attendants">
        <users>
          <user id="1002" type="pointer"/>
          <user id="1003" type="pointer"/>
        </users>
      </group>
</groups>

dialplan/default/mydialplan.xml: putting call on fifo
<extension name="sales_fifo_in">
      <condition field="destination_number" expression="^sales$" />
  <action application="answer"/>
    <action application="set" data="fifo_chime_freq=30"/>
<action application="set" data="fifo_orbit_dialplan=XML"/>
 <action application="set" data="fifo_orbit_context=default"/>
 <action application="set" data="fifo_caller_exit_key=0"/>
 <action application="set" data="fifo_caller_exit_to_orbit=true"/>
          <action application="set"
data="fifo_music=$${base_dir}/sounds/music/8000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav"/>
 <action application="fifo" data="sales_fifo@$${domain} in undef"/>
    </extension>

It used to work but nows it only sends one call at a time, even with the
other users available. I also tried replacing the groups for users but no
luck.

Thanks

On Fri, Jun 3, 2011 at 12:53 PM, Michael Collins <msc at freeswitch.org> wrote:

> Also, check out this thread, particularly the comments from Anthony:
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2011-March/070588.html
>
> If you had a really old version of FIFO and updated then you may have
> crossed over from when caller ID was not the default to the current state
> where it is the default.
>
> -MC
>
> On Thu, Jun 2, 2011 at 2:02 PM, Joao Leme <joaocarlosleme at gmail.com>wrote:
>
>> Hi there,
>> No change nor updates have been made in months. Everything was working
>> great but now FIFO is not ringing the extensions while the first call is not
>> HUNG UP...meaning, first call to go to FIFO rings the agents, and one
>> person answer...when the second call comes in, it goes to fifo queue (caller
>> listens the music) but no one is aware of the call, nor even the agents that
>> are available. Only after the first call hangs up the 2nd goes in. I tried
>> troubleshooting but no help.
>>
>> What I also noticed is that before, to display the caller ID on my sip
>> client I had to use:
>>
>> <action application="set"
>> data="origination_caller_id_number=${caller_id_number}"/>
>> <action application="set"
>> data="origination_caller_id_name=${caller_id_name}"/>
>>
>> but now, while testing, the caller id was showing when calling
>> fifo without setting those parameters, but before it would just show the
>> fifo queue name.
>>
>> All I can say is that I can't figure out why and how to fix these behavior
>> if I have done no change to FreeSWITCH nor the config files. I also tried
>> downloading the latest GIT and got the same response.
>>
>> THANKS,
>> John
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From yungwei at resolvity.com  Sat Jun  4 02:35:48 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Fri, 3 Jun 2011 18:35:48 -0400
Subject: [Freeswitch-users] A DTMF issue
Message-ID: <33095823FD21DF429B481B5163264B7950AC585DEE@VMBX102.ihostexchange.net>

Hi,

When testing the following javascript program, I notice a problem that it takes longer and longer for freeswitch to respond to DTMF input for some reason.
Please refer to http://pastebin.freeswitch.org/16435 for debug logs.
Notice that the difference in time between any adjacent pairs of the following is increasing as time goes by. Am I missing something here? 
	switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
	switch_rtp.c:3302 RTP RECV DTMF 2:960
Thanks.

Here's the javascript program:
var dtmf_digits = "";


function on_dtmf_28(session, type, digits, arg)
{
    if (type == "dtmf")
    {

        dtmf_digits = digits.digit;

        console_log("dtmf_digits=" + dtmf_digits + "\n");

        return false; // barge-in / done
    }
    return true;
}

while (true)
{
        dtmf_digits = "";
        session.flushDigits();
        session.streamFile("/usr/local/freeswitch/sounds/long-prompt.wav", on_dtmf_28, false);

        if(dtmf_digits.length == 0)
        {
            /* no input */
            console_log("no input\n");
            session.speak('flite', 'kal', 'no input!', null);
        }
        else if(dtmf_digits == "1" || dtmf_digits == "2")
        {
                console_log("match "+dtmf_digits+"\n");
                session.speak('flite', 'kal', 'you pressed ' + dtmf_digits+ '!', null);
        }
        else
        {
            /* no match */
            console_log("no match\n");
            session.speak('flite', 'kal', "no match!", null);
        }

}


From steveayre at gmail.com  Sat Jun  4 02:46:20 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 3 Jun 2011 23:46:20 +0100
Subject: [Freeswitch-users] A DTMF issue
In-Reply-To: <33095823FD21DF429B481B5163264B7950AC585DEE@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950AC585DEE@VMBX102.ihostexchange.net>
Message-ID: <BANLkTi=vcvoLy=eU7f_=0bGCJ=2K2NjGfw@mail.gmail.com>

You should be checking session.ready() in that loop, as it is I think you'll
find the scripts are never ending when the channel is hung up.

streamFile/flushDigits probably return instantly, but then get called again
on the next loop iteration. The problem would get slightly worse after each
test call since there would be an extra copy of the script running.

Try replacing:
    while(true) {
with:
    while(session.ready()) {

-Steve


On 3 June 2011 23:35, Yungwei Chen <yungwei at resolvity.com> wrote:

> Hi,
>
> When testing the following javascript program, I notice a problem that it
> takes longer and longer for freeswitch to respond to DTMF input for some
> reason.
> Please refer to http://pastebin.freeswitch.org/16435 for debug logs.
> Notice that the difference in time between any adjacent pairs of the
> following is increasing as time goes by. Am I missing something here?
>        switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels
> 20ms
>        switch_rtp.c:3302 RTP RECV DTMF 2:960
> Thanks.
>
> Here's the javascript program:
> var dtmf_digits = "";
>
>
> function on_dtmf_28(session, type, digits, arg)
> {
>    if (type == "dtmf")
>    {
>
>        dtmf_digits = digits.digit;
>
>        console_log("dtmf_digits=" + dtmf_digits + "\n");
>
>        return false; // barge-in / done
>    }
>    return true;
> }
>
> while (true)
> {
>        dtmf_digits = "";
>        session.flushDigits();
>        session.streamFile("/usr/local/freeswitch/sounds/long-prompt.wav",
> on_dtmf_28, false);
>
>        if(dtmf_digits.length == 0)
>        {
>            /* no input */
>            console_log("no input\n");
>            session.speak('flite', 'kal', 'no input!', null);
>        }
>        else if(dtmf_digits == "1" || dtmf_digits == "2")
>        {
>                console_log("match "+dtmf_digits+"\n");
>                session.speak('flite', 'kal', 'you pressed ' + dtmf_digits+
> '!', null);
>        }
>        else
>        {
>            /* no match */
>            console_log("no match\n");
>            session.speak('flite', 'kal', "no match!", null);
>        }
>
> }
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Sat Jun  4 02:48:30 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 3 Jun 2011 23:48:30 +0100
Subject: [Freeswitch-users] Need help interpreting fs_cli debug messages
In-Reply-To: <BANLkTingE3hspOajHyB12aszr68rgit_KA@mail.gmail.com>
References: <BANLkTingE3hspOajHyB12aszr68rgit_KA@mail.gmail.com>
Message-ID: <BANLkTikXOb909Sg0w=sY0pbetL2MwVnYeQ@mail.gmail.com>

The debug log and siptrace would be more useful than the sofia stack
debugging...

Looks like it's failing to authenticate, which is likely bad credentials or
the configuration not quite matching what the provider needs.

-Steve




On 3 June 2011 23:04, Roman Gelfand <rgelfand2 at gmail.com> wrote:

> I was wondering if somebody could help interpret the registration issue.
>
> +OK log level  [7]
> freeswitch at internal> nua: nh_create_handle: entering
> nua: nua_handle_bind: entering
> 2011-06-02 13:35:18.401497 [NOTICE] sofia_reg.c:367 Registering
> ###########-callcentric.com
> nua: nua_register: entering
> nua(0x873dc88): sent signal r_register
> nua: nua_stack_set_params: entering
> soa_clone(static::0x86d2e30, 0x86fba08, 0x873dc88) called
> soa_set_params(static::0x8dd11008, ...) called
> soa_set_params(static::0x8dd11008, ...) called
> nua(0x873dc88): adding register usage
> nta_leg_tcreate(0x8dd16398)
> nta: selecting scheme sip
> tport_tsend(0x86afae0) tpn = */XX.XX.XXX.XXX:5060
> tport_resolve addrinfo = XX.XX.XXX.XXX:5060
> tport_by_addrinfo(0x86afae0): not found by name */XX.XX.XXX.XXX:5060
> tport_vsend returned 639
> nta: sent REGISTER (13181995) to */XX.XX.XXX.XXX:5060
> tport_pend(0x86afae0): pending 0x8dd0a588 for udp/XX.XX.XXX.XXX:5080
> (already 0)
> nta: timer set to 32000 ms
> nta: timer shortened to 1000 ms
> tport_wakeup_pri(0x86afae0): events IN
> tport_recv_event(0x86afae0)
> tport_recv_iovec(0x86afae0) msg 0x8dd08cf8 from
> (udp/XX.XX.XXX.XXX:5080) has 526 bytes, veclen = 1
> tport_deliver(0x86afae0): msg 0x8dd08cf8 (526 bytes) from
> udp/XX.XX.XXX.XXX:5080/sip next=(nil)
> nta: received 407 Proxy Authentication Required for REGISTER (13181995)
> nta: 407 Proxy Authentication Required is going to a transaction
> nta_outgoing: RTT is 135.683 ms
> tport_release(0x86afae0): 0x8dd0a588 by 0x8dd009d8 with 0x8dd08cf8
> auth_digest_challenge_get(): got 7
> nta: outgoing_free(0x8dd009d8)
> nua: nua_application_event: entering
> nua: nua_authenticate: entering
> nua(0x873dc88): sent signal r_authenticate
> nua: nua_handle_magic: entering
> auth_digest_a1() has A1 = MD5(###########:callcentric.com:passwd) =
> 8dddd49339918097d09e22dd7b15c38b
> A2 = MD5(REGISTER:sip:XX.XX.XXX.XXX:5060)
> auth_response: 9f45cc31d0328fdf18a7e4d3b547e3ea =
>
> MD5(8dddd49339918097d09e22dd7b15c38b:dbcc6c9095c35c4be730ec1110f970e5:db5f9cddc4d0d297aa85cc660f20595a)
> (qop=NONE)
> nta: selecting scheme sip
> tport_tsend(0x86afae0) tpn = */XX.XX.XXX.XXX:5060
> tport_resolve addrinfo = XX.XX.XXX.XXX:5060
> tport_by_addrinfo(0x86afae0): not found by name */XX.XX.XXX.XXX:5060
> tport_vsend returned 859
> nta: sent REGISTER (13181996) to */XX.XX.XXX.XXX:5060
> tport_pend(0x86afae0): pending 0x8dd08cf8 for udp/XX.XX.XXX.XXX:5080
> (already 0)
> tport_wakeup_pri(0x86afae0): events IN
> tport_recv_event(0x86afae0)
> tport_recv_iovec(0x86afae0) msg 0x8dd1ae88 from
> (udp/XX.XX.XXX.XXX:5080) has 526 bytes, veclen = 1
> tport_deliver(0x86afae0): msg 0x8dd1ae88 (526 bytes) from
> udp/XX.XX.XXX.XXX:5080/sip next=(nil)
> nta: received 407 Proxy Authentication Required for REGISTER (13181996)
> nta: 407 Proxy Authentication Required is going to a transaction
> nta_outgoing: RTT is 127.055 ms
> tport_release(0x86afae0): 0x8dd08cf8 by 0x8dd08a98 with 0x8dd1ae88
> auth_digest_challenge_get(): got 7
> auth_digest_a1() has A1 = MD5(###########:callcentric.com:passwd) =
> 8dddd49339918097d09e22dd7b15c38b
> A2 = MD5(REGISTER:sip:XX.XX.XXX.XXX:5060)
> auth_response: c0b3965ec27e2e3386277645ae14f324 =
>
> MD5(8dddd49339918097d09e22dd7b15c38b:c76385015930949803ca8b0946f81d0e:db5f9cddc4d0d297aa85cc660f20595a)
> (qop=NONE)
> nta: selecting scheme sip
> tport_tsend(0x86afae0) tpn = */XX.XX.XXX.XXX:5060
> tport_resolve addrinfo = XX.XX.XXX.XXX:5060
> tport_by_addrinfo(0x86afae0): not found by name */XX.XX.XXX.XXX:5060
> tport_vsend returned 859
> nta: sent REGISTER (13181997) to */XX.XX.XXX.XXX:5060
> tport_pend(0x86afae0): pending 0x8dd19ea8 for udp/XX.XX.XXX.XXX:5080
> (already 0)
> nta: outgoing_free(0x8dd08a98)
> nua: nua_application_event: entering
> nua: nua_handle_magic: entering
> tport_wakeup_pri(0x86afae0): events IN
> tport_recv_event(0x86afae0)
> tport_recv_iovec(0x86afae0) msg 0x8dd1ae88 from
> (udp/XX.XX.XXX.XXX:5080) has 526 bytes, veclen = 1
> tport_deliver(0x86afae0): msg 0x8dd1ae88 (526 bytes) from
> udp/XX.XX.XXX.XXX:5080/sip next=(nil)
> nta: received 407 Proxy Authentication Required for REGISTER (13181997)
> nta: 407 Proxy Authentication Required is going to a transaction
> nta_outgoing: RTT is 139.175 ms
> tport_release(0x86afae0): 0x8dd19ea8 by 0x8dd18790 with 0x8dd1ae88
> auth_digest_challenge_get(): got 7
> auth_digest_a1() has A1 = MD5(###########:callcentric.com:passwd) =
> 8dddd49339918097d09e22dd7b15c38b
> A2 = MD5(REGISTER:sip:XX.XX.XXX.XXX:5060)
> auth_response: f014cc64fa858ec3615b2b1caab196a6 =
>
> MD5(8dddd49339918097d09e22dd7b15c38b:274bba14cbca0384c30f227da6c42c35:db5f9cddc4d0d297aa85cc660f20595a)
> (qop=NONE)
> nta: selecting scheme sip
> tport_tsend(0x86afae0) tpn = */XX.XX.XXX.XXX:5060
> tport_resolve addrinfo = XX.XX.XXX.XXX:5060
> tport_by_addrinfo(0x86afae0): not found by name */XX.XX.XXX.XXX:5060
> tport_vsend returned 859
> nta: sent REGISTER (13181998) to */XX.XX.XXX.XXX:5060
> tport_pend(0x86afae0): pending 0x8dd18f80 for udp/XX.XX.XXX.XXX:5080
> (already 0)
> nta: outgoing_free(0x8dd18790)
> nua: nua_application_event: entering
> nua: nua_handle_magic: entering
> tport_wakeup_pri(0x86afae0): events IN
> tport_recv_event(0x86afae0)
> tport_recv_iovec(0x86afae0) msg 0x8dd1ae88 from
> (udp/XX.XX.XXX.XXX:5080) has 526 bytes, veclen = 1
> tport_deliver(0x86afae0): msg 0x8dd1ae88 (526 bytes) from
> udp/XX.XX.XXX.XXX:5080/sip next=(nil)
> nta: received 407 Proxy Authentication Required for REGISTER (13181998)
> nta: 407 Proxy Authentication Required is going to a transaction
> nta_outgoing: RTT is 114.044 ms
> tport_release(0x86afae0): 0x8dd18f80 by 0x8dd16e10 with 0x8dd1ae88
> nta: outgoing_free(0x8dd16e10)
> nua(0x873dc88): removing register usage
> nta_leg_destroy(0x8dd16398)
> nua: nua_application_event: entering
> nua: nua_authenticate: entering
> nua(0x873dc88): sent signal r_authenticate
> nua: nua_handle_magic: entering
> nua: nua_application_event: entering
> nua: nua_handle_destroy: entering
> nua(0x873dc88): sent signal r_destroy
> nua: nua_handle_magic: entering
> nua: nua_handle_bind: entering
> nua: nua_handle_destroy: entering
> nta_leg_destroy((nil))
> soa_destroy(static::0x8dd11008) called
> nta: timer not set
> 2011-06-02 13:36:19.875422 [WARNING] sofia_reg.c:401 Timeout
> Registering ###########-callcentric.com
> nua: nua_handle_destroy: entering
> 2011-06-02 13:36:20.881513 [WARNING] sofia_reg.c:425
> ###########-callcentric.com Failed Registration [0], setting retry to
> 30 seconds.
>
>
> Thanks in advance
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sireeps at gmail.com  Sat Jun  4 07:14:38 2011
From: sireeps at gmail.com (Kamen)
Date: Fri, 3 Jun 2011 20:14:38 -0700 (PDT)
Subject: [Freeswitch-users] collectInput does not call callback function
In-Reply-To: <4de2ae9b.0b610e0a.3fd0.1fe0@mx.google.com>
References: <1306601304658-6414679.post@n2.nabble.com>
	<4de166bd.81320e0a.7135.7e06@mx.google.com>
	<1306683515927-6416704.post@n2.nabble.com>
	<4de2ae9b.0b610e0a.3fd0.1fe0@mx.google.com>
Message-ID: <1307157278537-6438355.post@n2.nabble.com>

I have exhausted all possible combinations for the javascript function
collectInput paramaters. Function is not called.  Although the timeout works
and it waits before continues.  Could it be that DTMF is not getting
through?  And accordingly the function is not called. What could be the
possible cause? 

Thanks

Sergei Kamen

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/collectInput-does-not-call-callback-function-tp6414679p6438355.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From gmaruzz at gmail.com  Sat Jun  4 09:39:07 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 4 Jun 2011 07:39:07 +0200
Subject: [Freeswitch-users] mod_skypopen is awesome issue with demo-ivr
	conference
In-Reply-To: <BANLkTinOpAXeV3wWMS=dHekQ-rw_d9urxA@mail.gmail.com>
References: <BANLkTik8hpzwjo-Susi6-wFDAzPiDZ-Jyw@mail.gmail.com>
	<BANLkTimu3GAitdDLvk=tcU+ZYUuXAjKCrw@mail.gmail.com>
	<BANLkTimmW0ZcGW2CwRkH+iQy-tQXHbCstA@mail.gmail.com>
	<BANLkTi=ULE8PKLOJRhhZkah71pZu_M0UqQ@mail.gmail.com>
	<BANLkTinOpAXeV3wWMS=dHekQ-rw_d9urxA@mail.gmail.com>
Message-ID: <BANLkTi=vDmeZ1rnYno6V6Yg20LmrEVY1Hg@mail.gmail.com>

You need to set the variable
'skype_add_outband_dtmf_also_when_bridged' to true before bridging.

This is my /usr/local/freeswitch/conf/dialplan/default/02_conf_via_skype.xml :

<include>
    <extension name="conf_via_skype">
      <condition field="destination_number" expression="^5004$">
        <action application="set"
data="skype_add_outband_dtmf_also_when_bridged=true"/>
        <action application="set" data="ringback=$${us-ring}"/>
        <action application="sleep" data="500"/>
        <action application="bridge"
data="sofia/${use_profile}/888 at conference.freeswitch.org"/>
      </condition>
    </extension>
</include>

You can copy and paste it in the same file.

You then will edit the file
/usr/local/freeswitch/conf/ivr_menus/demo_ivr.xml and substitute the
line:

<entry action="menu-exec-app" digits="1" param="bridge
sofia/$${domain}/888 at conference.freeswitch.org"/>

with the line:

<entry action="menu-exec-app" digits="1" param="transfer 5004 XML default"/>

and it will work as you expect, both with inbound skype and with inbound sip.

-giovanni


On Fri, Jun 3, 2011 at 11:08 PM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> As per the wiki page, and as is clearly written in the console and the
> logfile, if an incomong skype call is bridged to an outbound call, the
> dtmf is by default passed only in band (eg: as audio).
> If you want to have it working with the fs developer conference,
> create an extension that goes there and add to it the variable that's
> in the wiki page (can't remember the name, is something like
> "dtmf-outband-also-when-bridged=true").
> Please check the wiki page for the variable name.
> Reason for this is that the in band dtmf cannot be shut up, and often
> is detected by the remote party. But in this particular case of the fs
> developer conference, the remote party is setup to not detect inband
> in sip (or anyway, it does not detect it).
> Long story short: create an extension that sets that variable and goes
> to the fs conference and you'll be all set.
> Tomorrow I'll post here an example from my dialplan.
> -giovanni
>
> On 6/3/11, Michael Collins <msc at freeswitch.org> wrote:
>> Try transferring to a local conference on your machine and see what happens.
>> Watch the fs_cli and see if the DTMFs show up or not. From there we can see
>> what's up...
>>
>> -MC
>>
>> On Fri, Jun 3, 2011 at 11:10 AM, Henry Huang
>> <b_ball_henry at hotmail.com>wrote:
>>
>>> But the DTMF press worked very well while I was under the Demo IVR, even
>>> when I press "1001" I was able to get connected to extension 1001. It
>>> didn't
>>> work only after I was transferred to the conference selection.
>>>
>>> Henry
>>>
>>> On Sat, Jun 4, 2011 at 12:35 AM, Michael Collins
>>> <msc at freeswitch.org>wrote:
>>>
>>>> The monkeys have stopped screaming for everyone. :( I have been looking
>>>> for an alternative sound file for this but haven't found anything I
>>>> really
>>>> like.
>>>>
>>>> Not sure about the DTMF, but from what I understand there have been
>>>> issues
>>>> with sending DTMFs from the Skype client. I haven't tried it myself so I
>>>> will defer to those who have.
>>>>
>>>> -MC
>>>>
>>>> On Fri, Jun 3, 2011 at 9:19 AM, Henry Huang
>>>> <b_ball_henry at hotmail.com>wrote:
>>>>
>>>>> Hi:
>>>>>
>>>>> The new interactive installation for mod_skypopen is a piece of cake.
>>>>> Made live much easier. I was so excited and tried it out right after the
>>>>> installation was done. But I found a few issues calling the demo-ivr
>>>>> from
>>>>> skyTpe client.
>>>>>
>>>>> 1. The screaming monkey does not scream anymore.
>>>>> 2. The conference that connects to the freeswitch.org doesn't respond to
>>>>> dtmf pressed on the skype client. Ever since the second client is
>>>>> connected
>>>>> to the conference, the conference tell the client it's muted. So I tried
>>>>> push 0 and any other key to see if I get to unmute. But no, non of the
>>>>> key
>>>>> press works.
>>>>>
>>>>> If anyone have previous experiences on these and fixed them, please do
>>>>> share your methods.
>>>>>
>>>>> Sound quality with the new virtual sound drive is very good.
>>>>>
>>>>> Thanks,
>>>>>
>>>>> Henry Huang
>>>>> US: +1(818)6885508 | ??(Taiwan): +886 933847619
>>>>> Contact Me [image:
>>>>> LinkedIn]<http://www.linkedin.com/profile/view?id=4590380> [image:
>>>>> Facebook] <http://www.facebook.com/profile.php?id=595148342> [image:
>>>>> Twitter] <http://twitter.com/unicsolution>
>>>>> IM [image: Google Talk] red_rain_seven at gmail.com [image:
>>>>> Skype]unicsolution [image:
>>>>> MSN] b_ball_henry at hotmail.com
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> --
> Sent from my mobile device
>
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From u2nsam at gmail.com  Sat Jun  4 10:06:39 2011
From: u2nsam at gmail.com (Sam)
Date: Sat, 4 Jun 2011 11:36:39 +0530
Subject: [Freeswitch-users] domain
In-Reply-To: <4DE8A680.1080102@tagnet.ru>
References: <BANLkTimO7+Sg=8cJjdfRr-1wEKgbbZDrcA@mail.gmail.com>
	<4DE89971.9010205@tagnet.ru>
	<BANLkTimOZ-yxbNXFiOr519FoR=9euOZLcg@mail.gmail.com>
	<4DE8A680.1080102@tagnet.ru>
Message-ID: <BANLkTinkghJTnEw16ED7PfnbmUBAPk=9dA@mail.gmail.com>

Hello,

After doing this ,

<action application="set" data="domain
name=${regex(${sip_from_uri}|^.*@(.*)$|%1)}" inline="true"/>

i get domain name = xyz.com

Now how can i use the fundas of identified domain and using below to execute
.

<domain name="xyz.com">
<condition field="destination_number" expression="^(1001)$">
 <action application="voicemail" data="default $${domain name}
${sip_to_host}"/>
</domain>



Is that possible.


Regards
Sam


On Fri, Jun 3, 2011 at 2:46 PM, Boris Kovalenko <boris at tagnet.ru> wrote:

>  Hello!
>
>     There is a variable named ${sip_from_user}. You may do something like
>
> <action application="set" data=
> "from_domain=${regex(${sip_from_user}|^.*@(.*)$|%1)}"<from_domain=$%7Bregex%28$%7Bsip_from_user%7D%7C%5E.*@%28.*%29$%7C%251%29%7D>inline="true"/>
> <action application="transfer" data="${from_domain} XML domain_context"/>
>
>
> The call would be coming from outside to FS and not within , so the call
> would be routed from public to different domains ,
>
> and after that following the doc
> http://wiki.freeswitch.org/wiki/Multi-tenant
>
> Here how can we recognized from the header on which domain is it ?  like if
> we get an from header as 1001 at xyz.com
> how can we recognized the domain and route the call accordingly ... when an
> outside system sends a call via acl .
>
> regards
> Sam
>
> On Fri, Jun 3, 2011 at 1:51 PM, Boris Kovalenko <boris at tagnet.ru> wrote:
>
>>  Hello!
>>
>>     What do You mean? You may (for example) set the domain variable for a
>> user and test it inside condition.
>>
>>  Hello,
>>
>> I have a user registered by domain (1001 at xyz.com) and when the invite
>> comes to my FS server it should recognize the domain and route
>> the calls accordingly to the group <domain name="xyz.com">  ...
>> </domain> .
>> The call would be entering via the acl to public.xml. Here i want to route
>> the call according to domain names, how will i do that.
>>
>> Any suggestions...
>>
>>
>> Regards
>> Sam
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> --
>> Regards,
>> Boris
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> Regards,
> Boris
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From boris at tagnet.ru  Sat Jun  4 10:51:19 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sat, 04 Jun 2011 12:51:19 +0600
Subject: [Freeswitch-users] domain
In-Reply-To: <BANLkTinkghJTnEw16ED7PfnbmUBAPk=9dA@mail.gmail.com>
References: <BANLkTimO7+Sg=8cJjdfRr-1wEKgbbZDrcA@mail.gmail.com>	<4DE89971.9010205@tagnet.ru>	<BANLkTimOZ-yxbNXFiOr519FoR=9euOZLcg@mail.gmail.com>	<4DE8A680.1080102@tagnet.ru>
	<BANLkTinkghJTnEw16ED7PfnbmUBAPk=9dA@mail.gmail.com>
Message-ID: <4DE9D5E7.7030800@tagnet.ru>

Hello!

     No, You can't use it this way. There are 2 possibilites:
1. You have one context (dialplan) and use something like

<extension name="ext_01">
<condition field="${domain_name}" expression="^xyz.com$"/>
<condition filed="destination_number" expression="^(1001)$">
<action application="voicemail" data="default $${domain name} 
${sip_to_host}"/>
</condition>
</extension>

2. I prefer the way to have separate contexts (dialplans) for each 
domain. So in the main context You do something like:
<action application="tranfser" data="${destination_number} XML 
context_${domain_name}"/>
And after that You know that is context_${domain_name} (context_xyz.com 
as in your example) You have o
> Hello,
>
> After doing this ,
>
> <action application="set" data="domain 
> name=${regex(${sip_from_uri}|^.*@(.*)$|%1)}" inline="true"/>
>
> i get domain name = xyz.com <http://xyz.com>
>
> Now how can i use the fundas of identified domain and using below to 
> execute .
>
> <domain name="xyz.com <http://xyz.com>">
> <condition field="destination_number" expression="^(1001)$">
> <action application="voicemail" data="default $${domain name} 
> ${sip_to_host}"/>
> </domain>
>
>
>
> Is that possible.
>
>
> Regards
> Sam
>
>
> On Fri, Jun 3, 2011 at 2:46 PM, Boris Kovalenko <boris at tagnet.ru 
> <mailto:boris at tagnet.ru>> wrote:
>
>     Hello!
>
>         There is a variable named ${sip_from_user}. You may do
>     something like
>
>     <action application="set"
>     data="from_domain=${regex(${sip_from_user}|^.*@(.*)$|%1)}"
>     <mailto:from_domain=$%7Bregex%28$%7Bsip_from_user%7D%7C%5E.*@%28.*%29$%7C%251%29%7D>
>     inline="true"/>
>     <action application="transfer" data="${from_domain} XML
>     domain_context"/>
>
>
>>     The call would be coming from outside to FS and not within , so
>>     the call would be routed from public to different domains ,
>>
>>     and after that following the doc
>>     http://wiki.freeswitch.org/wiki/Multi-tenant
>>
>>     Here how can we recognized from the header on which domain is it
>>     ?  like if we get an from header as 1001 at xyz.com
>>     <mailto:1001 at xyz.com>
>>     how can we recognized the domain and route the call accordingly
>>     ... when an outside system sends a call via acl .
>>
>>     regards
>>     Sam
>>
>>     On Fri, Jun 3, 2011 at 1:51 PM, Boris Kovalenko <boris at tagnet.ru
>>     <mailto:boris at tagnet.ru>> wrote:
>>
>>         Hello!
>>
>>             What do You mean? You may (for example) set the domain
>>         variable for a user and test it inside condition.
>>
>>>         Hello,
>>>
>>>         I have a user registered by domain (1001 at xyz.com
>>>         <mailto:1001 at xyz.com>) and when the invite comes to my FS
>>>         server it should recognize the domain and route
>>>         the calls accordingly to the group <domain name="xyz.com
>>>         <http://xyz.com>">  ... </domain> .
>>>         The call would be entering via the acl to public.xml. Here i
>>>         want to route the call according to domain names, how will i
>>>         do that.
>>>
>>>         Any suggestions...
>>>
>>>
>>>         Regards
>>>         Sam
>>>
>>>
>>>
>>>         _______________________________________________
>>>         FreeSWITCH-users mailing list
>>>         FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>         http://www.freeswitch.org
>>
>>
>>         -- 
>>         Regards,
>>         Boris
>>
>>
>>
>>         _______________________________________________
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>     -- 
>     Regards,
>     Boris
>
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Regards,
Boris


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From b_ball_henry at hotmail.com  Sat Jun  4 19:48:43 2011
From: b_ball_henry at hotmail.com (Henry Huang)
Date: Sat, 4 Jun 2011 23:48:43 +0800
Subject: [Freeswitch-users] mod_skypopen is awesome issue with demo-ivr
	conference
In-Reply-To: <BANLkTi=vDmeZ1rnYno6V6Yg20LmrEVY1Hg@mail.gmail.com>
References: <BANLkTik8hpzwjo-Susi6-wFDAzPiDZ-Jyw@mail.gmail.com>
	<BANLkTimu3GAitdDLvk=tcU+ZYUuXAjKCrw@mail.gmail.com>
	<BANLkTimmW0ZcGW2CwRkH+iQy-tQXHbCstA@mail.gmail.com>
	<BANLkTi=ULE8PKLOJRhhZkah71pZu_M0UqQ@mail.gmail.com>
	<BANLkTinOpAXeV3wWMS=dHekQ-rw_d9urxA@mail.gmail.com>
	<BANLkTi=vDmeZ1rnYno6V6Yg20LmrEVY1Hg@mail.gmail.com>
Message-ID: <BANLkTi=rg+jPAJ1ZYgnrODLE78w0CpuP1A@mail.gmail.com>

Giovanni:

Yeh, I saw the section regarding sending dtmf after bridge later. ?got too
excited once the installation was done and not finish reading the whole
thing. Thanks for making things more clear here. I will put your example to
the wiki if it's not already there.

By the way, is there a place where DTMF is explained in more detail of what
are the differences between rfc2833, inband, and info? For my understanding,
I think both inband and rfc2833 are transmitted via the RTP correct? and sip
info is by the SIP signal?

Thanks,



Henry Huang
US: +1(818)6885508 | ??(Taiwan): +886 933847619
Contact Me [image:
LinkedIn]<http://www.linkedin.com/profile/view?id=4590380> [image:
Facebook] <http://www.facebook.com/profile.php?id=595148342> [image:
Twitter] <http://twitter.com/unicsolution>
IM [image: Google Talk] red_rain_seven at gmail.com [image: Skype] unicsolution
[image: MSN] b_ball_henry at hotmail.com


On Sat, Jun 4, 2011 at 1:39 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> You need to set the variable
> 'skype_add_outband_dtmf_also_when_bridged' to true before bridging.
>
> This is my
> /usr/local/freeswitch/conf/dialplan/default/02_conf_via_skype.xml :
>
> <include>
>    <extension name="conf_via_skype">
>      <condition field="destination_number" expression="^5004$">
>        <action application="set"
> data="skype_add_outband_dtmf_also_when_bridged=true"/>
>        <action application="set" data="ringback=$${us-ring}"/>
>        <action application="sleep" data="500"/>
>        <action application="bridge"
> data="sofia/${use_profile}/888 at conference.freeswitch.org"/>
>      </condition>
>    </extension>
> </include>
>
> You can copy and paste it in the same file.
>
> You then will edit the file
> /usr/local/freeswitch/conf/ivr_menus/demo_ivr.xml and substitute the
> line:
>
> <entry action="menu-exec-app" digits="1" param="bridge
> sofia/$${domain}/888 at conference.freeswitch.org"/>
>
> with the line:
>
> <entry action="menu-exec-app" digits="1" param="transfer 5004 XML
> default"/>
>
> and it will work as you expect, both with inbound skype and with inbound
> sip.
>
> -giovanni
>
>
> On Fri, Jun 3, 2011 at 11:08 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
> > As per the wiki page, and as is clearly written in the console and the
> > logfile, if an incomong skype call is bridged to an outbound call, the
> > dtmf is by default passed only in band (eg: as audio).
> > If you want to have it working with the fs developer conference,
> > create an extension that goes there and add to it the variable that's
> > in the wiki page (can't remember the name, is something like
> > "dtmf-outband-also-when-bridged=true").
> > Please check the wiki page for the variable name.
> > Reason for this is that the in band dtmf cannot be shut up, and often
> > is detected by the remote party. But in this particular case of the fs
> > developer conference, the remote party is setup to not detect inband
> > in sip (or anyway, it does not detect it).
> > Long story short: create an extension that sets that variable and goes
> > to the fs conference and you'll be all set.
> > Tomorrow I'll post here an example from my dialplan.
> > -giovanni
> >
> > On 6/3/11, Michael Collins <msc at freeswitch.org> wrote:
> >> Try transferring to a local conference on your machine and see what
> happens.
> >> Watch the fs_cli and see if the DTMFs show up or not. From there we can
> see
> >> what's up...
> >>
> >> -MC
> >>
> >> On Fri, Jun 3, 2011 at 11:10 AM, Henry Huang
> >> <b_ball_henry at hotmail.com>wrote:
> >>
> >>> But the DTMF press worked very well while I was under the Demo IVR,
> even
> >>> when I press "1001" I was able to get connected to extension 1001. It
> >>> didn't
> >>> work only after I was transferred to the conference selection.
> >>>
> >>> Henry
> >>>
> >>> On Sat, Jun 4, 2011 at 12:35 AM, Michael Collins
> >>> <msc at freeswitch.org>wrote:
> >>>
> >>>> The monkeys have stopped screaming for everyone. :( I have been
> looking
> >>>> for an alternative sound file for this but haven't found anything I
> >>>> really
> >>>> like.
> >>>>
> >>>> Not sure about the DTMF, but from what I understand there have been
> >>>> issues
> >>>> with sending DTMFs from the Skype client. I haven't tried it myself so
> I
> >>>> will defer to those who have.
> >>>>
> >>>> -MC
> >>>>
> >>>> On Fri, Jun 3, 2011 at 9:19 AM, Henry Huang
> >>>> <b_ball_henry at hotmail.com>wrote:
> >>>>
> >>>>> Hi:
> >>>>>
> >>>>> The new interactive installation for mod_skypopen is a piece of cake.
> >>>>> Made live much easier. I was so excited and tried it out right after
> the
> >>>>> installation was done. But I found a few issues calling the demo-ivr
> >>>>> from
> >>>>> skyTpe client.
> >>>>>
> >>>>> 1. The screaming monkey does not scream anymore.
> >>>>> 2. The conference that connects to the freeswitch.org doesn't
> respond to
> >>>>> dtmf pressed on the skype client. Ever since the second client is
> >>>>> connected
> >>>>> to the conference, the conference tell the client it's muted. So I
> tried
> >>>>> push 0 and any other key to see if I get to unmute. But no, non of
> the
> >>>>> key
> >>>>> press works.
> >>>>>
> >>>>> If anyone have previous experiences on these and fixed them, please
> do
> >>>>> share your methods.
> >>>>>
> >>>>> Sound quality with the new virtual sound drive is very good.
> >>>>>
> >>>>> Thanks,
> >>>>>
> >>>>> Henry Huang
> >>>>> US: +1(818)6885508 | ??(Taiwan): +886 933847619
> >>>>> Contact Me [image:
> >>>>> LinkedIn]<http://www.linkedin.com/profile/view?id=4590380> [image:
> >>>>> Facebook] <http://www.facebook.com/profile.php?id=595148342> [image:
> >>>>> Twitter] <http://twitter.com/unicsolution>
> >>>>> IM [image: Google Talk] red_rain_seven at gmail.com [image:
> >>>>> Skype]unicsolution [image:
> >>>>> MSN] b_ball_henry at hotmail.com
> >>>>>
> >>>>>
> >>>>> _______________________________________________
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>>
> >>>>>
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>
> >
> > --
> > Sent from my mobile device
> >
> > Sincerely,
> >
> > Giovanni Maruzzelli
> > Cell : +39-347-2665618
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From boris at tagnet.ru  Sat Jun  4 20:05:45 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sat, 04 Jun 2011 22:05:45 +0600
Subject: [Freeswitch-users] Strage E1 PCI card
Message-ID: <4DEA57D9.1030103@tagnet.ru>

Hello!

     I found strange PCI E1 card marked as DIALOGIC DM/IP0821A-E1-120. 
No drivers, no manuals. It is possible to get it working with FreeSwitch?

-- 
Regards,
Boris




From curriegrad2004 at gmail.com  Sat Jun  4 20:09:24 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 4 Jun 2011 09:09:24 -0700
Subject: [Freeswitch-users] Strage E1 PCI card
In-Reply-To: <4DEA57D9.1030103@tagnet.ru>
References: <4DEA57D9.1030103@tagnet.ru>
Message-ID: <BANLkTi=hzqkEf1tiOh0RN0bw++vL_Zy=DQ@mail.gmail.com>

Haven't found any drivers out there that might support this. You're on
your own if you do go ahead.

On Sat, Jun 4, 2011 at 9:05 AM, Boris Kovalenko <boris at tagnet.ru> wrote:
> Hello!
>
> ? ? I found strange PCI E1 card marked as DIALOGIC DM/IP0821A-E1-120.
> No drivers, no manuals. It is possible to get it working with FreeSwitch?
>
> --
> Regards,
> Boris
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From kahn at vestec.com  Sat Jun  4 18:57:43 2011
From: kahn at vestec.com (Kashif Kahn)
Date: Sat, 04 Jun 2011 10:57:43 -0400
Subject: [Freeswitch-users] Speech Recognition: Australian English
Message-ID: <4DEA47E7.8010306@vestec.com>

Hello Everyone,

Vestec ASR engine now supports speech recognition in Australian English. 
(We also support American English, Indian English, and Chinese 
Mandarin). A starter kit is available for $25: 
http://www.vestec.com/products

Please note that Vestec offers the best deal around for enabling 
sophisticated speech recognition with command-and-control type IVR 
applications by offering a high accuracy, standards based speech engine 
at a fraction of the cost of conventional ASR vendors. Feel free to 
contact me with any questions or concerns.

Regards,
-Kashif



From educs13 at yahoo.com.br  Sat Jun  4 19:45:50 2011
From: educs13 at yahoo.com.br (educs13)
Date: Sat, 4 Jun 2011 08:45:50 -0700 (PDT)
Subject: [Freeswitch-users] Lua ASR TTS
Message-ID: <1307202350274-6439449.post@n2.nabble.com>

Hi,

I was trying to run the 'Examples directory lua asr tts'
(http://wiki.freeswitch.org/wiki/Examples_directory_lua_asr_tts) in Ubuntu
10.04/32 bits, but I had some problems with the 'session:streamFile()'.

If I say something when the prompt is still playing, the audio stucks
indefinitely. It seems that the 'break' command isn't doing all its job. I
guess that maybe the 'break' is just stop feeding the audio player with new
samples, but it is not stopping the player. So, this player keeps reading
and playing the same samples continuously ...

I really don't know if my guess makes sense and as I don't know C very well,
I don't know where to look for a solution in the source code. I Hope you
could help me.

Thanks,
Eduardo



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Lua-ASR-TTS-tp6439449p6439449.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From dujinfang at gmail.com  Sun Jun  5 03:24:36 2011
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 5 Jun 2011 07:24:36 +0800
Subject: [Freeswitch-users] Speech Recognition: Australian English
In-Reply-To: <4DEA47E7.8010306@vestec.com>
References: <4DEA47E7.8010306@vestec.com>
Message-ID: <5391553A31334E0587B99203F8E78C96@gmail.com>

Interested in dose it offer trial version?

-- 
Seven Du
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj: http://www.freeswitch.org.cn
Sent with Sparrow
On Saturday, June 4, 2011 at 10:57 PM, Kashif Kahn wrote: 
> Hello Everyone,
> 
> Vestec ASR engine now supports speech recognition in Australian English. 
> (We also support American English, Indian English, and Chinese 
> Mandarin). A starter kit is available for $25: 
> http://www.vestec.com/products
> 
> Please note that Vestec offers the best deal around for enabling 
> sophisticated speech recognition with command-and-control type IVR 
> applications by offering a high accuracy, standards based speech engine 
> at a fraction of the cost of conventional ASR vendors. Feel free to 
> contact me with any questions or concerns.
> 
> Regards,
> -Kashif
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
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From b_ball_henry at hotmail.com  Sun Jun  5 06:28:34 2011
From: b_ball_henry at hotmail.com (Henry Huang)
Date: Sun, 5 Jun 2011 10:28:34 +0800
Subject: [Freeswitch-users] mod_skypopen is awesome issue with demo-ivr
	conference
In-Reply-To: <BANLkTi=vDmeZ1rnYno6V6Yg20LmrEVY1Hg@mail.gmail.com>
References: <BANLkTik8hpzwjo-Susi6-wFDAzPiDZ-Jyw@mail.gmail.com>
	<BANLkTimu3GAitdDLvk=tcU+ZYUuXAjKCrw@mail.gmail.com>
	<BANLkTimmW0ZcGW2CwRkH+iQy-tQXHbCstA@mail.gmail.com>
	<BANLkTi=ULE8PKLOJRhhZkah71pZu_M0UqQ@mail.gmail.com>
	<BANLkTinOpAXeV3wWMS=dHekQ-rw_d9urxA@mail.gmail.com>
	<BANLkTi=vDmeZ1rnYno6V6Yg20LmrEVY1Hg@mail.gmail.com>
Message-ID: <BANLkTik6E5WtjTbVkBjRv3ykMo8rtnAB2A@mail.gmail.com>

One thing I didn't see it covered in the wiki. Is the skype client we use
for the mod_skypopen having the risk of being used as a "supernode"? If so,
is there ways to disable it?

Thanks,

Henry

On Sat, Jun 4, 2011 at 1:39 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> You need to set the variable
> 'skype_add_outband_dtmf_also_when_bridged' to true before bridging.
>
> This is my
> /usr/local/freeswitch/conf/dialplan/default/02_conf_via_skype.xml :
>
> <include>
>    <extension name="conf_via_skype">
>      <condition field="destination_number" expression="^5004$">
>        <action application="set"
> data="skype_add_outband_dtmf_also_when_bridged=true"/>
>        <action application="set" data="ringback=$${us-ring}"/>
>        <action application="sleep" data="500"/>
>        <action application="bridge"
> data="sofia/${use_profile}/888 at conference.freeswitch.org"/>
>      </condition>
>    </extension>
> </include>
>
> You can copy and paste it in the same file.
>
> You then will edit the file
> /usr/local/freeswitch/conf/ivr_menus/demo_ivr.xml and substitute the
> line:
>
> <entry action="menu-exec-app" digits="1" param="bridge
> sofia/$${domain}/888 at conference.freeswitch.org"/>
>
> with the line:
>
> <entry action="menu-exec-app" digits="1" param="transfer 5004 XML
> default"/>
>
> and it will work as you expect, both with inbound skype and with inbound
> sip.
>
> -giovanni
>
>
> On Fri, Jun 3, 2011 at 11:08 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
> > As per the wiki page, and as is clearly written in the console and the
> > logfile, if an incomong skype call is bridged to an outbound call, the
> > dtmf is by default passed only in band (eg: as audio).
> > If you want to have it working with the fs developer conference,
> > create an extension that goes there and add to it the variable that's
> > in the wiki page (can't remember the name, is something like
> > "dtmf-outband-also-when-bridged=true").
> > Please check the wiki page for the variable name.
> > Reason for this is that the in band dtmf cannot be shut up, and often
> > is detected by the remote party. But in this particular case of the fs
> > developer conference, the remote party is setup to not detect inband
> > in sip (or anyway, it does not detect it).
> > Long story short: create an extension that sets that variable and goes
> > to the fs conference and you'll be all set.
> > Tomorrow I'll post here an example from my dialplan.
> > -giovanni
> >
> > On 6/3/11, Michael Collins <msc at freeswitch.org> wrote:
> >> Try transferring to a local conference on your machine and see what
> happens.
> >> Watch the fs_cli and see if the DTMFs show up or not. From there we can
> see
> >> what's up...
> >>
> >> -MC
> >>
> >> On Fri, Jun 3, 2011 at 11:10 AM, Henry Huang
> >> <b_ball_henry at hotmail.com>wrote:
> >>
> >>> But the DTMF press worked very well while I was under the Demo IVR,
> even
> >>> when I press "1001" I was able to get connected to extension 1001. It
> >>> didn't
> >>> work only after I was transferred to the conference selection.
> >>>
> >>> Henry
> >>>
> >>> On Sat, Jun 4, 2011 at 12:35 AM, Michael Collins
> >>> <msc at freeswitch.org>wrote:
> >>>
> >>>> The monkeys have stopped screaming for everyone. :( I have been
> looking
> >>>> for an alternative sound file for this but haven't found anything I
> >>>> really
> >>>> like.
> >>>>
> >>>> Not sure about the DTMF, but from what I understand there have been
> >>>> issues
> >>>> with sending DTMFs from the Skype client. I haven't tried it myself so
> I
> >>>> will defer to those who have.
> >>>>
> >>>> -MC
> >>>>
> >>>> On Fri, Jun 3, 2011 at 9:19 AM, Henry Huang
> >>>> <b_ball_henry at hotmail.com>wrote:
> >>>>
> >>>>> Hi:
> >>>>>
> >>>>> The new interactive installation for mod_skypopen is a piece of cake.
> >>>>> Made live much easier. I was so excited and tried it out right after
> the
> >>>>> installation was done. But I found a few issues calling the demo-ivr
> >>>>> from
> >>>>> skyTpe client.
> >>>>>
> >>>>> 1. The screaming monkey does not scream anymore.
> >>>>> 2. The conference that connects to the freeswitch.org doesn't
> respond to
> >>>>> dtmf pressed on the skype client. Ever since the second client is
> >>>>> connected
> >>>>> to the conference, the conference tell the client it's muted. So I
> tried
> >>>>> push 0 and any other key to see if I get to unmute. But no, non of
> the
> >>>>> key
> >>>>> press works.
> >>>>>
> >>>>> If anyone have previous experiences on these and fixed them, please
> do
> >>>>> share your methods.
> >>>>>
> >>>>> Sound quality with the new virtual sound drive is very good.
> >>>>>
> >>>>> Thanks,
> >>>>>
> >>>>> Henry Huang
> >>>>> US: +1(818)6885508 | ??(Taiwan): +886 933847619
> >>>>> Contact Me [image:
> >>>>> LinkedIn]<http://www.linkedin.com/profile/view?id=4590380> [image:
> >>>>> Facebook] <http://www.facebook.com/profile.php?id=595148342> [image:
> >>>>> Twitter] <http://twitter.com/unicsolution>
> >>>>> IM [image: Google Talk] red_rain_seven at gmail.com [image:
> >>>>> Skype]unicsolution [image:
> >>>>> MSN] b_ball_henry at hotmail.com
> >>>>>
> >>>>>
> >>>>> _______________________________________________
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>>
> >>>>>
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>
> >
> > --
> > Sent from my mobile device
> >
> > Sincerely,
> >
> > Giovanni Maruzzelli
> > Cell : +39-347-2665618
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gmaruzz at celliax.org  Sun Jun  5 10:21:32 2011
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sun, 5 Jun 2011 08:21:32 +0200
Subject: [Freeswitch-users] mod_skypopen is awesome issue with demo-ivr
	conference
In-Reply-To: <BANLkTik6E5WtjTbVkBjRv3ykMo8rtnAB2A@mail.gmail.com>
References: <BANLkTik8hpzwjo-Susi6-wFDAzPiDZ-Jyw@mail.gmail.com>
	<BANLkTimu3GAitdDLvk=tcU+ZYUuXAjKCrw@mail.gmail.com>
	<BANLkTimmW0ZcGW2CwRkH+iQy-tQXHbCstA@mail.gmail.com>
	<BANLkTi=ULE8PKLOJRhhZkah71pZu_M0UqQ@mail.gmail.com>
	<BANLkTinOpAXeV3wWMS=dHekQ-rw_d9urxA@mail.gmail.com>
	<BANLkTi=vDmeZ1rnYno6V6Yg20LmrEVY1Hg@mail.gmail.com>
	<BANLkTik6E5WtjTbVkBjRv3ykMo8rtnAB2A@mail.gmail.com>
Message-ID: <BANLkTik2jtQsrihVJGwrUz_KS3G3WSMrzQ@mail.gmail.com>

If you're behind a nat or a firewall you can't be supernode.
If you're directly on the internet (eg: you have a real routable IP
number) on the machine, and you are not behind a firewall (eg: you
have all incoming ports open) then you *may* become a supernode.
Btw: google is your friend, those are informations easily available
with a search for "skype supernode".
-giovanni

On 6/5/11, Henry Huang <b_ball_henry at hotmail.com> wrote:
> One thing I didn't see it covered in the wiki. Is the skype client we use
> for the mod_skypopen having the risk of being used as a "supernode"? If so,
> is there ways to disable it?
>
> Thanks,
>
> Henry
>
> On Sat, Jun 4, 2011 at 1:39 PM, Giovanni Maruzzelli
> <gmaruzz at gmail.com>wrote:
>
>> You need to set the variable
>> 'skype_add_outband_dtmf_also_when_bridged' to true before bridging.
>>
>> This is my
>> /usr/local/freeswitch/conf/dialplan/default/02_conf_via_skype.xml :
>>
>> <include>
>>    <extension name="conf_via_skype">
>>      <condition field="destination_number" expression="^5004$">
>>        <action application="set"
>> data="skype_add_outband_dtmf_also_when_bridged=true"/>
>>        <action application="set" data="ringback=$${us-ring}"/>
>>        <action application="sleep" data="500"/>
>>        <action application="bridge"
>> data="sofia/${use_profile}/888 at conference.freeswitch.org"/>
>>      </condition>
>>    </extension>
>> </include>
>>
>> You can copy and paste it in the same file.
>>
>> You then will edit the file
>> /usr/local/freeswitch/conf/ivr_menus/demo_ivr.xml and substitute the
>> line:
>>
>> <entry action="menu-exec-app" digits="1" param="bridge
>> sofia/$${domain}/888 at conference.freeswitch.org"/>
>>
>> with the line:
>>
>> <entry action="menu-exec-app" digits="1" param="transfer 5004 XML
>> default"/>
>>
>> and it will work as you expect, both with inbound skype and with inbound
>> sip.
>>
>> -giovanni
>>
>>
>> On Fri, Jun 3, 2011 at 11:08 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>> wrote:
>> > As per the wiki page, and as is clearly written in the console and the
>> > logfile, if an incomong skype call is bridged to an outbound call, the
>> > dtmf is by default passed only in band (eg: as audio).
>> > If you want to have it working with the fs developer conference,
>> > create an extension that goes there and add to it the variable that's
>> > in the wiki page (can't remember the name, is something like
>> > "dtmf-outband-also-when-bridged=true").
>> > Please check the wiki page for the variable name.
>> > Reason for this is that the in band dtmf cannot be shut up, and often
>> > is detected by the remote party. But in this particular case of the fs
>> > developer conference, the remote party is setup to not detect inband
>> > in sip (or anyway, it does not detect it).
>> > Long story short: create an extension that sets that variable and goes
>> > to the fs conference and you'll be all set.
>> > Tomorrow I'll post here an example from my dialplan.
>> > -giovanni
>> >
>> > On 6/3/11, Michael Collins <msc at freeswitch.org> wrote:
>> >> Try transferring to a local conference on your machine and see what
>> happens.
>> >> Watch the fs_cli and see if the DTMFs show up or not. From there we can
>> see
>> >> what's up...
>> >>
>> >> -MC
>> >>
>> >> On Fri, Jun 3, 2011 at 11:10 AM, Henry Huang
>> >> <b_ball_henry at hotmail.com>wrote:
>> >>
>> >>> But the DTMF press worked very well while I was under the Demo IVR,
>> even
>> >>> when I press "1001" I was able to get connected to extension 1001. It
>> >>> didn't
>> >>> work only after I was transferred to the conference selection.
>> >>>
>> >>> Henry
>> >>>
>> >>> On Sat, Jun 4, 2011 at 12:35 AM, Michael Collins
>> >>> <msc at freeswitch.org>wrote:
>> >>>
>> >>>> The monkeys have stopped screaming for everyone. :( I have been
>> looking
>> >>>> for an alternative sound file for this but haven't found anything I
>> >>>> really
>> >>>> like.
>> >>>>
>> >>>> Not sure about the DTMF, but from what I understand there have been
>> >>>> issues
>> >>>> with sending DTMFs from the Skype client. I haven't tried it myself
>> >>>> so
>> I
>> >>>> will defer to those who have.
>> >>>>
>> >>>> -MC
>> >>>>
>> >>>> On Fri, Jun 3, 2011 at 9:19 AM, Henry Huang
>> >>>> <b_ball_henry at hotmail.com>wrote:
>> >>>>
>> >>>>> Hi:
>> >>>>>
>> >>>>> The new interactive installation for mod_skypopen is a piece of
>> >>>>> cake.
>> >>>>> Made live much easier. I was so excited and tried it out right after
>> the
>> >>>>> installation was done. But I found a few issues calling the demo-ivr
>> >>>>> from
>> >>>>> skyTpe client.
>> >>>>>
>> >>>>> 1. The screaming monkey does not scream anymore.
>> >>>>> 2. The conference that connects to the freeswitch.org doesn't
>> respond to
>> >>>>> dtmf pressed on the skype client. Ever since the second client is
>> >>>>> connected
>> >>>>> to the conference, the conference tell the client it's muted. So I
>> tried
>> >>>>> push 0 and any other key to see if I get to unmute. But no, non of
>> the
>> >>>>> key
>> >>>>> press works.
>> >>>>>
>> >>>>> If anyone have previous experiences on these and fixed them, please
>> do
>> >>>>> share your methods.
>> >>>>>
>> >>>>> Sound quality with the new virtual sound drive is very good.
>> >>>>>
>> >>>>> Thanks,
>> >>>>>
>> >>>>> Henry Huang
>> >>>>> US: +1(818)6885508 | ??(Taiwan): +886 933847619
>> >>>>> Contact Me [image:
>> >>>>> LinkedIn]<http://www.linkedin.com/profile/view?id=4590380> [image:
>> >>>>> Facebook] <http://www.facebook.com/profile.php?id=595148342> [image:
>> >>>>> Twitter] <http://twitter.com/unicsolution>
>> >>>>> IM [image: Google Talk] red_rain_seven at gmail.com [image:
>> >>>>> Skype]unicsolution [image:
>> >>>>> MSN] b_ball_henry at hotmail.com
>> >>>>>
>> >>>>>
>> >>>>> _______________________________________________
>> >>>>> FreeSWITCH-users mailing list
>> >>>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>> http://www.freeswitch.org
>> >>>>>
>> >>>>>
>> >>>>
>> >>>> _______________________________________________
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>>
>> >>>>
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>>
>> >>
>> >
>> > --
>> > Sent from my mobile device
>> >
>> > Sincerely,
>> >
>> > Giovanni Maruzzelli
>> > Cell : +39-347-2665618
>> >
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From steveayre at gmail.com  Sun Jun  5 11:17:00 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 5 Jun 2011 08:17:00 +0100
Subject: [Freeswitch-users] mod_skypopen is awesome issue with demo-ivr
	conference
In-Reply-To: <BANLkTik2jtQsrihVJGwrUz_KS3G3WSMrzQ@mail.gmail.com>
References: <BANLkTik8hpzwjo-Susi6-wFDAzPiDZ-Jyw@mail.gmail.com>
	<BANLkTimu3GAitdDLvk=tcU+ZYUuXAjKCrw@mail.gmail.com>
	<BANLkTimmW0ZcGW2CwRkH+iQy-tQXHbCstA@mail.gmail.com>
	<BANLkTi=ULE8PKLOJRhhZkah71pZu_M0UqQ@mail.gmail.com>
	<BANLkTinOpAXeV3wWMS=dHekQ-rw_d9urxA@mail.gmail.com>
	<BANLkTi=vDmeZ1rnYno6V6Yg20LmrEVY1Hg@mail.gmail.com>
	<BANLkTik6E5WtjTbVkBjRv3ykMo8rtnAB2A@mail.gmail.com>
	<BANLkTik2jtQsrihVJGwrUz_KS3G3WSMrzQ@mail.gmail.com>
Message-ID: <00D2C8FA-7A2E-4066-B428-CE1A4B915744@gmail.com>

I think you might still be able to be a supernode behind nat if your router uses upnp... Not sure though.

Afaik the option to disable being a supernode was added on Windows in v3... No such option on Linux yet (v2).

Steve on iPhone

On 5 Jun 2011, at 07:21, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:

> If you're behind a nat or a firewall you can't be supernode.
> If you're directly on the internet (eg: you have a real routable IP
> number) on the machine, and you are not behind a firewall (eg: you
> have all incoming ports open) then you *may* become a supernode.
> Btw: google is your friend, those are informations easily available
> with a search for "skype supernode".
> -giovanni
> 
> On 6/5/11, Henry Huang <b_ball_henry at hotmail.com> wrote:
>> One thing I didn't see it covered in the wiki. Is the skype client we use
>> for the mod_skypopen having the risk of being used as a "supernode"? If so,
>> is there ways to disable it?
>> 
>> Thanks,
>> 
>> Henry
>> 
>> On Sat, Jun 4, 2011 at 1:39 PM, Giovanni Maruzzelli
>> <gmaruzz at gmail.com>wrote:
>> 
>>> You need to set the variable
>>> 'skype_add_outband_dtmf_also_when_bridged' to true before bridging.
>>> 
>>> This is my
>>> /usr/local/freeswitch/conf/dialplan/default/02_conf_via_skype.xml :
>>> 
>>> <include>
>>>   <extension name="conf_via_skype">
>>>     <condition field="destination_number" expression="^5004$">
>>>       <action application="set"
>>> data="skype_add_outband_dtmf_also_when_bridged=true"/>
>>>       <action application="set" data="ringback=$${us-ring}"/>
>>>       <action application="sleep" data="500"/>
>>>       <action application="bridge"
>>> data="sofia/${use_profile}/888 at conference.freeswitch.org"/>
>>>     </condition>
>>>   </extension>
>>> </include>
>>> 
>>> You can copy and paste it in the same file.
>>> 
>>> You then will edit the file
>>> /usr/local/freeswitch/conf/ivr_menus/demo_ivr.xml and substitute the
>>> line:
>>> 
>>> <entry action="menu-exec-app" digits="1" param="bridge
>>> sofia/$${domain}/888 at conference.freeswitch.org"/>
>>> 
>>> with the line:
>>> 
>>> <entry action="menu-exec-app" digits="1" param="transfer 5004 XML
>>> default"/>
>>> 
>>> and it will work as you expect, both with inbound skype and with inbound
>>> sip.
>>> 
>>> -giovanni
>>> 
>>> 
>>> On Fri, Jun 3, 2011 at 11:08 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>>> wrote:
>>>> As per the wiki page, and as is clearly written in the console and the
>>>> logfile, if an incomong skype call is bridged to an outbound call, the
>>>> dtmf is by default passed only in band (eg: as audio).
>>>> If you want to have it working with the fs developer conference,
>>>> create an extension that goes there and add to it the variable that's
>>>> in the wiki page (can't remember the name, is something like
>>>> "dtmf-outband-also-when-bridged=true").
>>>> Please check the wiki page for the variable name.
>>>> Reason for this is that the in band dtmf cannot be shut up, and often
>>>> is detected by the remote party. But in this particular case of the fs
>>>> developer conference, the remote party is setup to not detect inband
>>>> in sip (or anyway, it does not detect it).
>>>> Long story short: create an extension that sets that variable and goes
>>>> to the fs conference and you'll be all set.
>>>> Tomorrow I'll post here an example from my dialplan.
>>>> -giovanni
>>>> 
>>>> On 6/3/11, Michael Collins <msc at freeswitch.org> wrote:
>>>>> Try transferring to a local conference on your machine and see what
>>> happens.
>>>>> Watch the fs_cli and see if the DTMFs show up or not. From there we can
>>> see
>>>>> what's up...
>>>>> 
>>>>> -MC
>>>>> 
>>>>> On Fri, Jun 3, 2011 at 11:10 AM, Henry Huang
>>>>> <b_ball_henry at hotmail.com>wrote:
>>>>> 
>>>>>> But the DTMF press worked very well while I was under the Demo IVR,
>>> even
>>>>>> when I press "1001" I was able to get connected to extension 1001. It
>>>>>> didn't
>>>>>> work only after I was transferred to the conference selection.
>>>>>> 
>>>>>> Henry
>>>>>> 
>>>>>> On Sat, Jun 4, 2011 at 12:35 AM, Michael Collins
>>>>>> <msc at freeswitch.org>wrote:
>>>>>> 
>>>>>>> The monkeys have stopped screaming for everyone. :( I have been
>>> looking
>>>>>>> for an alternative sound file for this but haven't found anything I
>>>>>>> really
>>>>>>> like.
>>>>>>> 
>>>>>>> Not sure about the DTMF, but from what I understand there have been
>>>>>>> issues
>>>>>>> with sending DTMFs from the Skype client. I haven't tried it myself
>>>>>>> so
>>> I
>>>>>>> will defer to those who have.
>>>>>>> 
>>>>>>> -MC
>>>>>>> 
>>>>>>> On Fri, Jun 3, 2011 at 9:19 AM, Henry Huang
>>>>>>> <b_ball_henry at hotmail.com>wrote:
>>>>>>> 
>>>>>>>> Hi:
>>>>>>>> 
>>>>>>>> The new interactive installation for mod_skypopen is a piece of
>>>>>>>> cake.
>>>>>>>> Made live much easier. I was so excited and tried it out right after
>>> the
>>>>>>>> installation was done. But I found a few issues calling the demo-ivr
>>>>>>>> from
>>>>>>>> skyTpe client.
>>>>>>>> 
>>>>>>>> 1. The screaming monkey does not scream anymore.
>>>>>>>> 2. The conference that connects to the freeswitch.org doesn't
>>> respond to
>>>>>>>> dtmf pressed on the skype client. Ever since the second client is
>>>>>>>> connected
>>>>>>>> to the conference, the conference tell the client it's muted. So I
>>> tried
>>>>>>>> push 0 and any other key to see if I get to unmute. But no, non of
>>> the
>>>>>>>> key
>>>>>>>> press works.
>>>>>>>> 
>>>>>>>> If anyone have previous experiences on these and fixed them, please
>>> do
>>>>>>>> share your methods.
>>>>>>>> 
>>>>>>>> Sound quality with the new virtual sound drive is very good.
>>>>>>>> 
>>>>>>>> Thanks,
>>>>>>>> 
>>>>>>>> Henry Huang
>>>>>>>> US: +1(818)6885508 | ??(Taiwan): +886 933847619
>>>>>>>> Contact Me [image:
>>>>>>>> LinkedIn]<http://www.linkedin.com/profile/view?id=4590380> [image:
>>>>>>>> Facebook] <http://www.facebook.com/profile.php?id=595148342> [image:
>>>>>>>> Twitter] <http://twitter.com/unicsolution>
>>>>>>>> IM [image: Google Talk] red_rain_seven at gmail.com [image:
>>>>>>>> Skype]unicsolution [image:
>>>>>>>> MSN] b_ball_henry at hotmail.com
>>>>>>>> 
>>>>>>>> 
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>> 
>>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>> 
>>>> 
>>>> --
>>>> Sent from my mobile device
>>>> 
>>>> Sincerely,
>>>> 
>>>> Giovanni Maruzzelli
>>>> Cell : +39-347-2665618
>>>> 
>>> 
>>> 
>>> 
>>> --
>>> Sincerely,
>>> 
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>> 
> 
> -- 
> Sent from my mobile device
> 
> Sincerely,
> 
> Giovanni Maruzzelli
> Cell : +39-347-2665618
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From b_ball_henry at hotmail.com  Sun Jun  5 17:39:33 2011
From: b_ball_henry at hotmail.com (Henry Huang)
Date: Sun, 5 Jun 2011 21:39:33 +0800
Subject: [Freeswitch-users] mod_skypopen is awesome issue with demo-ivr
	conference
In-Reply-To: <BANLkTik2jtQsrihVJGwrUz_KS3G3WSMrzQ@mail.gmail.com>
References: <BANLkTik8hpzwjo-Susi6-wFDAzPiDZ-Jyw@mail.gmail.com>
	<BANLkTimu3GAitdDLvk=tcU+ZYUuXAjKCrw@mail.gmail.com>
	<BANLkTimmW0ZcGW2CwRkH+iQy-tQXHbCstA@mail.gmail.com>
	<BANLkTi=ULE8PKLOJRhhZkah71pZu_M0UqQ@mail.gmail.com>
	<BANLkTinOpAXeV3wWMS=dHekQ-rw_d9urxA@mail.gmail.com>
	<BANLkTi=vDmeZ1rnYno6V6Yg20LmrEVY1Hg@mail.gmail.com>
	<BANLkTik6E5WtjTbVkBjRv3ykMo8rtnAB2A@mail.gmail.com>
	<BANLkTik2jtQsrihVJGwrUz_KS3G3WSMrzQ@mail.gmail.com>
Message-ID: <BANLkTi=zGa4JEHeOKXPj2XWWbAnX+_x4AA@mail.gmail.com>

Sorry, I didn't make it clear. I did search google and found "general"
information of skype being behind NAT or firewall. But I was looking for
something that is capable of turning that feature off. So I tossed the
questions here hoping people having done integration with Skype would know
better.

I also thought of some feature that I am not sure if I should posted here or
open a new thread. I was thinking of integrating the mod_skypopen with
mod_digaling or some other sort so that when the skype client receive
messages, we can transfer it to a designated XMPP client with a 2 way text
design in mind.


Thanks,

Henry

On Sun, Jun 5, 2011 at 2:21 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> If you're behind a nat or a firewall you can't be supernode.
> If you're directly on the internet (eg: you have a real routable IP
> number) on the machine, and you are not behind a firewall (eg: you
> have all incoming ports open) then you *may* become a supernode.
> Btw: google is your friend, those are informations easily available
> with a search for "skype supernode".
> -giovanni
>
> On 6/5/11, Henry Huang <b_ball_henry at hotmail.com> wrote:
> > One thing I didn't see it covered in the wiki. Is the skype client we use
> > for the mod_skypopen having the risk of being used as a "supernode"? If
> so,
> > is there ways to disable it?
> >
> > Thanks,
> >
> > Henry
> >
> > On Sat, Jun 4, 2011 at 1:39 PM, Giovanni Maruzzelli
> > <gmaruzz at gmail.com>wrote:
> >
> >> You need to set the variable
> >> 'skype_add_outband_dtmf_also_when_bridged' to true before bridging.
> >>
> >> This is my
> >> /usr/local/freeswitch/conf/dialplan/default/02_conf_via_skype.xml :
> >>
> >> <include>
> >>    <extension name="conf_via_skype">
> >>      <condition field="destination_number" expression="^5004$">
> >>        <action application="set"
> >> data="skype_add_outband_dtmf_also_when_bridged=true"/>
> >>        <action application="set" data="ringback=$${us-ring}"/>
> >>        <action application="sleep" data="500"/>
> >>        <action application="bridge"
> >> data="sofia/${use_profile}/888 at conference.freeswitch.org"/>
> >>      </condition>
> >>    </extension>
> >> </include>
> >>
> >> You can copy and paste it in the same file.
> >>
> >> You then will edit the file
> >> /usr/local/freeswitch/conf/ivr_menus/demo_ivr.xml and substitute the
> >> line:
> >>
> >> <entry action="menu-exec-app" digits="1" param="bridge
> >> sofia/$${domain}/888 at conference.freeswitch.org"/>
> >>
> >> with the line:
> >>
> >> <entry action="menu-exec-app" digits="1" param="transfer 5004 XML
> >> default"/>
> >>
> >> and it will work as you expect, both with inbound skype and with inbound
> >> sip.
> >>
> >> -giovanni
> >>
> >>
> >> On Fri, Jun 3, 2011 at 11:08 PM, Giovanni Maruzzelli <gmaruzz at gmail.com
> >
> >> wrote:
> >> > As per the wiki page, and as is clearly written in the console and the
> >> > logfile, if an incomong skype call is bridged to an outbound call, the
> >> > dtmf is by default passed only in band (eg: as audio).
> >> > If you want to have it working with the fs developer conference,
> >> > create an extension that goes there and add to it the variable that's
> >> > in the wiki page (can't remember the name, is something like
> >> > "dtmf-outband-also-when-bridged=true").
> >> > Please check the wiki page for the variable name.
> >> > Reason for this is that the in band dtmf cannot be shut up, and often
> >> > is detected by the remote party. But in this particular case of the fs
> >> > developer conference, the remote party is setup to not detect inband
> >> > in sip (or anyway, it does not detect it).
> >> > Long story short: create an extension that sets that variable and goes
> >> > to the fs conference and you'll be all set.
> >> > Tomorrow I'll post here an example from my dialplan.
> >> > -giovanni
> >> >
> >> > On 6/3/11, Michael Collins <msc at freeswitch.org> wrote:
> >> >> Try transferring to a local conference on your machine and see what
> >> happens.
> >> >> Watch the fs_cli and see if the DTMFs show up or not. From there we
> can
> >> see
> >> >> what's up...
> >> >>
> >> >> -MC
> >> >>
> >> >> On Fri, Jun 3, 2011 at 11:10 AM, Henry Huang
> >> >> <b_ball_henry at hotmail.com>wrote:
> >> >>
> >> >>> But the DTMF press worked very well while I was under the Demo IVR,
> >> even
> >> >>> when I press "1001" I was able to get connected to extension 1001.
> It
> >> >>> didn't
> >> >>> work only after I was transferred to the conference selection.
> >> >>>
> >> >>> Henry
> >> >>>
> >> >>> On Sat, Jun 4, 2011 at 12:35 AM, Michael Collins
> >> >>> <msc at freeswitch.org>wrote:
> >> >>>
> >> >>>> The monkeys have stopped screaming for everyone. :( I have been
> >> looking
> >> >>>> for an alternative sound file for this but haven't found anything I
> >> >>>> really
> >> >>>> like.
> >> >>>>
> >> >>>> Not sure about the DTMF, but from what I understand there have been
> >> >>>> issues
> >> >>>> with sending DTMFs from the Skype client. I haven't tried it myself
> >> >>>> so
> >> I
> >> >>>> will defer to those who have.
> >> >>>>
> >> >>>> -MC
> >> >>>>
> >> >>>> On Fri, Jun 3, 2011 at 9:19 AM, Henry Huang
> >> >>>> <b_ball_henry at hotmail.com>wrote:
> >> >>>>
> >> >>>>> Hi:
> >> >>>>>
> >> >>>>> The new interactive installation for mod_skypopen is a piece of
> >> >>>>> cake.
> >> >>>>> Made live much easier. I was so excited and tried it out right
> after
> >> the
> >> >>>>> installation was done. But I found a few issues calling the
> demo-ivr
> >> >>>>> from
> >> >>>>> skyTpe client.
> >> >>>>>
> >> >>>>> 1. The screaming monkey does not scream anymore.
> >> >>>>> 2. The conference that connects to the freeswitch.org doesn't
> >> respond to
> >> >>>>> dtmf pressed on the skype client. Ever since the second client is
> >> >>>>> connected
> >> >>>>> to the conference, the conference tell the client it's muted. So I
> >> tried
> >> >>>>> push 0 and any other key to see if I get to unmute. But no, non of
> >> the
> >> >>>>> key
> >> >>>>> press works.
> >> >>>>>
> >> >>>>> If anyone have previous experiences on these and fixed them,
> please
> >> do
> >> >>>>> share your methods.
> >> >>>>>
> >> >>>>> Sound quality with the new virtual sound drive is very good.
> >> >>>>>
> >> >>>>> Thanks,
> >> >>>>>
> >> >>>>> Henry Huang
> >> >>>>> US: +1(818)6885508 | ??(Taiwan): +886 933847619
> >> >>>>> Contact Me [image:
> >> >>>>> LinkedIn]<http://www.linkedin.com/profile/view?id=4590380>
> [image:
> >> >>>>> Facebook] <http://www.facebook.com/profile.php?id=595148342>
> [image:
> >> >>>>> Twitter] <http://twitter.com/unicsolution>
> >> >>>>> IM [image: Google Talk] red_rain_seven at gmail.com [image:
> >> >>>>> Skype]unicsolution [image:
> >> >>>>> MSN] b_ball_henry at hotmail.com
> >> >>>>>
> >> >>>>>
> >> >>>>> _______________________________________________
> >> >>>>> FreeSWITCH-users mailing list
> >> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>>> UNSUBSCRIBE:
> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>>>> http://www.freeswitch.org
> >> >>>>>
> >> >>>>>
> >> >>>>
> >> >>>> _______________________________________________
> >> >>>> FreeSWITCH-users mailing list
> >> >>>> FreeSWITCH-users at lists.freeswitch.org
> >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>> UNSUBSCRIBE:
> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>>> http://www.freeswitch.org
> >> >>>>
> >> >>>>
> >> >>>
> >> >>> _______________________________________________
> >> >>> FreeSWITCH-users mailing list
> >> >>> FreeSWITCH-users at lists.freeswitch.org
> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> UNSUBSCRIBE:
> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> http://www.freeswitch.org
> >> >>>
> >> >>>
> >> >>
> >> >
> >> > --
> >> > Sent from my mobile device
> >> >
> >> > Sincerely,
> >> >
> >> > Giovanni Maruzzelli
> >> > Cell : +39-347-2665618
> >> >
> >>
> >>
> >>
> >> --
> >> Sincerely,
> >>
> >> Giovanni Maruzzelli
> >> Cell : +39-347-2665618
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
>
> --
> Sent from my mobile device
>
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gmaruzz at celliax.org  Sun Jun  5 18:38:31 2011
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sun, 5 Jun 2011 16:38:31 +0200
Subject: [Freeswitch-users] mod_skypopen is awesome issue with demo-ivr
	conference
In-Reply-To: <BANLkTi=zGa4JEHeOKXPj2XWWbAnX+_x4AA@mail.gmail.com>
References: <BANLkTik8hpzwjo-Susi6-wFDAzPiDZ-Jyw@mail.gmail.com>
	<BANLkTimu3GAitdDLvk=tcU+ZYUuXAjKCrw@mail.gmail.com>
	<BANLkTimmW0ZcGW2CwRkH+iQy-tQXHbCstA@mail.gmail.com>
	<BANLkTi=ULE8PKLOJRhhZkah71pZu_M0UqQ@mail.gmail.com>
	<BANLkTinOpAXeV3wWMS=dHekQ-rw_d9urxA@mail.gmail.com>
	<BANLkTi=vDmeZ1rnYno6V6Yg20LmrEVY1Hg@mail.gmail.com>
	<BANLkTik6E5WtjTbVkBjRv3ykMo8rtnAB2A@mail.gmail.com>
	<BANLkTik2jtQsrihVJGwrUz_KS3G3WSMrzQ@mail.gmail.com>
	<BANLkTi=zGa4JEHeOKXPj2XWWbAnX+_x4AA@mail.gmail.com>
Message-ID: <BANLkTin5fL=8-mVmM+kv0w9U_qYDWY0eaQ@mail.gmail.com>

:)
again, read the wiki page.
Messages use the CHAT standard freeswitch interface, same used by
dingaling ans sofia.
So, you can simply make an application that routes messages between
skype, google and sip, eg, using the ESL.
-giovanni

On 6/5/11, Henry Huang <b_ball_henry at hotmail.com> wrote:
> Sorry, I didn't make it clear. I did search google and found "general"
> information of skype being behind NAT or firewall. But I was looking for
> something that is capable of turning that feature off. So I tossed the
> questions here hoping people having done integration with Skype would know
> better.
>
> I also thought of some feature that I am not sure if I should posted here or
> open a new thread. I was thinking of integrating the mod_skypopen with
> mod_digaling or some other sort so that when the skype client receive
> messages, we can transfer it to a designated XMPP client with a 2 way text
> design in mind.
>
>
> Thanks,
>
> Henry
>
> On Sun, Jun 5, 2011 at 2:21 PM, Giovanni Maruzzelli
> <gmaruzz at celliax.org>wrote:
>
>> If you're behind a nat or a firewall you can't be supernode.
>> If you're directly on the internet (eg: you have a real routable IP
>> number) on the machine, and you are not behind a firewall (eg: you
>> have all incoming ports open) then you *may* become a supernode.
>> Btw: google is your friend, those are informations easily available
>> with a search for "skype supernode".
>> -giovanni
>>
>> On 6/5/11, Henry Huang <b_ball_henry at hotmail.com> wrote:
>> > One thing I didn't see it covered in the wiki. Is the skype client we
>> > use
>> > for the mod_skypopen having the risk of being used as a "supernode"? If
>> so,
>> > is there ways to disable it?
>> >
>> > Thanks,
>> >
>> > Henry
>> >
>> > On Sat, Jun 4, 2011 at 1:39 PM, Giovanni Maruzzelli
>> > <gmaruzz at gmail.com>wrote:
>> >
>> >> You need to set the variable
>> >> 'skype_add_outband_dtmf_also_when_bridged' to true before bridging.
>> >>
>> >> This is my
>> >> /usr/local/freeswitch/conf/dialplan/default/02_conf_via_skype.xml :
>> >>
>> >> <include>
>> >>    <extension name="conf_via_skype">
>> >>      <condition field="destination_number" expression="^5004$">
>> >>        <action application="set"
>> >> data="skype_add_outband_dtmf_also_when_bridged=true"/>
>> >>        <action application="set" data="ringback=$${us-ring}"/>
>> >>        <action application="sleep" data="500"/>
>> >>        <action application="bridge"
>> >> data="sofia/${use_profile}/888 at conference.freeswitch.org"/>
>> >>      </condition>
>> >>    </extension>
>> >> </include>
>> >>
>> >> You can copy and paste it in the same file.
>> >>
>> >> You then will edit the file
>> >> /usr/local/freeswitch/conf/ivr_menus/demo_ivr.xml and substitute the
>> >> line:
>> >>
>> >> <entry action="menu-exec-app" digits="1" param="bridge
>> >> sofia/$${domain}/888 at conference.freeswitch.org"/>
>> >>
>> >> with the line:
>> >>
>> >> <entry action="menu-exec-app" digits="1" param="transfer 5004 XML
>> >> default"/>
>> >>
>> >> and it will work as you expect, both with inbound skype and with
>> >> inbound
>> >> sip.
>> >>
>> >> -giovanni
>> >>
>> >>
>> >> On Fri, Jun 3, 2011 at 11:08 PM, Giovanni Maruzzelli <gmaruzz at gmail.com
>> >
>> >> wrote:
>> >> > As per the wiki page, and as is clearly written in the console and
>> >> > the
>> >> > logfile, if an incomong skype call is bridged to an outbound call,
>> >> > the
>> >> > dtmf is by default passed only in band (eg: as audio).
>> >> > If you want to have it working with the fs developer conference,
>> >> > create an extension that goes there and add to it the variable that's
>> >> > in the wiki page (can't remember the name, is something like
>> >> > "dtmf-outband-also-when-bridged=true").
>> >> > Please check the wiki page for the variable name.
>> >> > Reason for this is that the in band dtmf cannot be shut up, and often
>> >> > is detected by the remote party. But in this particular case of the
>> >> > fs
>> >> > developer conference, the remote party is setup to not detect inband
>> >> > in sip (or anyway, it does not detect it).
>> >> > Long story short: create an extension that sets that variable and
>> >> > goes
>> >> > to the fs conference and you'll be all set.
>> >> > Tomorrow I'll post here an example from my dialplan.
>> >> > -giovanni
>> >> >
>> >> > On 6/3/11, Michael Collins <msc at freeswitch.org> wrote:
>> >> >> Try transferring to a local conference on your machine and see what
>> >> happens.
>> >> >> Watch the fs_cli and see if the DTMFs show up or not. From there we
>> can
>> >> see
>> >> >> what's up...
>> >> >>
>> >> >> -MC
>> >> >>
>> >> >> On Fri, Jun 3, 2011 at 11:10 AM, Henry Huang
>> >> >> <b_ball_henry at hotmail.com>wrote:
>> >> >>
>> >> >>> But the DTMF press worked very well while I was under the Demo IVR,
>> >> even
>> >> >>> when I press "1001" I was able to get connected to extension 1001.
>> It
>> >> >>> didn't
>> >> >>> work only after I was transferred to the conference selection.
>> >> >>>
>> >> >>> Henry
>> >> >>>
>> >> >>> On Sat, Jun 4, 2011 at 12:35 AM, Michael Collins
>> >> >>> <msc at freeswitch.org>wrote:
>> >> >>>
>> >> >>>> The monkeys have stopped screaming for everyone. :( I have been
>> >> looking
>> >> >>>> for an alternative sound file for this but haven't found anything
>> >> >>>> I
>> >> >>>> really
>> >> >>>> like.
>> >> >>>>
>> >> >>>> Not sure about the DTMF, but from what I understand there have
>> >> >>>> been
>> >> >>>> issues
>> >> >>>> with sending DTMFs from the Skype client. I haven't tried it
>> >> >>>> myself
>> >> >>>> so
>> >> I
>> >> >>>> will defer to those who have.
>> >> >>>>
>> >> >>>> -MC
>> >> >>>>
>> >> >>>> On Fri, Jun 3, 2011 at 9:19 AM, Henry Huang
>> >> >>>> <b_ball_henry at hotmail.com>wrote:
>> >> >>>>
>> >> >>>>> Hi:
>> >> >>>>>
>> >> >>>>> The new interactive installation for mod_skypopen is a piece of
>> >> >>>>> cake.
>> >> >>>>> Made live much easier. I was so excited and tried it out right
>> after
>> >> the
>> >> >>>>> installation was done. But I found a few issues calling the
>> demo-ivr
>> >> >>>>> from
>> >> >>>>> skyTpe client.
>> >> >>>>>
>> >> >>>>> 1. The screaming monkey does not scream anymore.
>> >> >>>>> 2. The conference that connects to the freeswitch.org doesn't
>> >> respond to
>> >> >>>>> dtmf pressed on the skype client. Ever since the second client is
>> >> >>>>> connected
>> >> >>>>> to the conference, the conference tell the client it's muted. So
>> >> >>>>> I
>> >> tried
>> >> >>>>> push 0 and any other key to see if I get to unmute. But no, non
>> >> >>>>> of
>> >> the
>> >> >>>>> key
>> >> >>>>> press works.
>> >> >>>>>
>> >> >>>>> If anyone have previous experiences on these and fixed them,
>> please
>> >> do
>> >> >>>>> share your methods.
>> >> >>>>>
>> >> >>>>> Sound quality with the new virtual sound drive is very good.
>> >> >>>>>
>> >> >>>>> Thanks,
>> >> >>>>>
>> >> >>>>> Henry Huang
>> >> >>>>> US: +1(818)6885508 | ??(Taiwan): +886 933847619
>> >> >>>>> Contact Me [image:
>> >> >>>>> LinkedIn]<http://www.linkedin.com/profile/view?id=4590380>
>> [image:
>> >> >>>>> Facebook] <http://www.facebook.com/profile.php?id=595148342>
>> [image:
>> >> >>>>> Twitter] <http://twitter.com/unicsolution>
>> >> >>>>> IM [image: Google Talk] red_rain_seven at gmail.com [image:
>> >> >>>>> Skype]unicsolution [image:
>> >> >>>>> MSN] b_ball_henry at hotmail.com
>> >> >>>>>
>> >> >>>>>
>> >> >>>>> _______________________________________________
>> >> >>>>> FreeSWITCH-users mailing list
>> >> >>>>> FreeSWITCH-users at lists.freeswitch.org
>> >> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>>>> UNSUBSCRIBE:
>> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>>>> http://www.freeswitch.org
>> >> >>>>>
>> >> >>>>>
>> >> >>>>
>> >> >>>> _______________________________________________
>> >> >>>> FreeSWITCH-users mailing list
>> >> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>>> UNSUBSCRIBE:
>> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>>> http://www.freeswitch.org
>> >> >>>>
>> >> >>>>
>> >> >>>
>> >> >>> _______________________________________________
>> >> >>> FreeSWITCH-users mailing list
>> >> >>> FreeSWITCH-users at lists.freeswitch.org
>> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>> UNSUBSCRIBE:
>> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>> http://www.freeswitch.org
>> >> >>>
>> >> >>>
>> >> >>
>> >> >
>> >> > --
>> >> > Sent from my mobile device
>> >> >
>> >> > Sincerely,
>> >> >
>> >> > Giovanni Maruzzelli
>> >> > Cell : +39-347-2665618
>> >> >
>> >>
>> >>
>> >>
>> >> --
>> >> Sincerely,
>> >>
>> >> Giovanni Maruzzelli
>> >> Cell : +39-347-2665618
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>>
>> --
>> Sent from my mobile device
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gustavo.espeche at easyipcall.com  Sun Jun  5 18:40:41 2011
From: gustavo.espeche at easyipcall.com (Gustavo Espeche)
Date: Sun, 05 Jun 2011 11:40:41 -0300
Subject: [Freeswitch-users] Sangoma Transcoding
Message-ID: <1307284841.2616.4.camel@gustavo-laptop>

Hi some one can do work sangoma d100 transcoding, i installed the card,
load the sangoma codec into freeswitch, but i can't do that the
freeswitch do a transcoding using sangoma card.
I configure my profile 
    <param name="inbound-late-negotiation" value="false"/>
    <param name="disable-transcoding" value="false"/>
if some one configure with success this card i appreciate a lot any
advice.
Best Regards.


-- 

Gustavo Espeche
EasyIpCall S.R.L.
www.easyipcall.com
Bv Mitre 517 24? E
Cordoba - Argentina
Te: +54 - 351 - 4280633



From steveayre at gmail.com  Sun Jun  5 20:04:47 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 5 Jun 2011 17:04:47 +0100
Subject: [Freeswitch-users] Sangoma Transcoding
In-Reply-To: <1307284841.2616.4.camel@gustavo-laptop>
References: <1307284841.2616.4.camel@gustavo-laptop>
Message-ID: <BANLkTin-P6Tyxw2=11qXLgiE_Rq+j1-yJA@mail.gmail.com>

Do you have mod_sangoma_codec loaded? If so, it'll automatically use the
card instead of CPU for transcoding for the supported codecs.

You should also avoid having the modules loaded that provide any of the
codecs that mod_sangoma_codec provides. i.e. do not load mod_g729,
mod_com_g729, mod_g7231, mod_amr, mod_ilbc.

The parameters you mentioned only affect how codec negotiation is performed,
they don't control transcoding and any settings for those parameter will
work with the Sandoma card. Rather importantly disable-transcoding is a bit
of a misnomer, it doesn't actually disable transcoding, just attempts to
force it not to be needed by forcing the aleg and bleg codecs to match.

-Steve




On 5 June 2011 15:40, Gustavo Espeche <gustavo.espeche at easyipcall.com>wrote:

> Hi some one can do work sangoma d100 transcoding, i installed the card,
> load the sangoma codec into freeswitch, but i can't do that the
> freeswitch do a transcoding using sangoma card.
> I configure my profile
>    <param name="inbound-late-negotiation" value="false"/>
>    <param name="disable-transcoding" value="false"/>
> if some one configure with success this card i appreciate a lot any
> advice.
> Best Regards.
>
>
> --
>
> Gustavo Espechese
> EasyIpCall S.R.L.
> www.easyipcall.com
> Bv Mitre 517 24? E
> Cordoba - Argentina
> Te: +54 - 351 - 4280633
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From moises.silva at gmail.com  Mon Jun  6 02:40:17 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Sun, 5 Jun 2011 18:40:17 -0400
Subject: [Freeswitch-users] Sangoma Transcoding
In-Reply-To: <BANLkTin-P6Tyxw2=11qXLgiE_Rq+j1-yJA@mail.gmail.com>
References: <1307284841.2616.4.camel@gustavo-laptop>
	<BANLkTin-P6Tyxw2=11qXLgiE_Rq+j1-yJA@mail.gmail.com>
Message-ID: <BANLkTikJcMXy-pTq2zjg9Mw8Se+AsGHhAg@mail.gmail.com>

On Sun, Jun 5, 2011 at 12:04 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Do you have mod_sangoma_codec loaded? If so, it'll automatically use the
> card instead of CPU for transcoding for the supported codecs.
>
> You should also avoid having the modules loaded that provide any of the
> codecs that mod_sangoma_codec provides. i.e. do not load mod_g729,
> mod_com_g729, mod_g7231, mod_amr, mod_ilbc.
>
>
Also, be aware, by default only G729 codec is registered when loading
mod_sangoma_codec. If you want other codecs you must enable them in
autload_configs/sangoma_codec.conf.xml

<param name="register" value="all"/>

Something like that would register all supported codecs by the card. As
stated by Steve you should not load conflicting modules, for example loading
mod_voipcodecs and registering G722 codec in sangoma_codec.conf.xml would
cause a conflict.

Moises Silva
Senior Software Engineer, Software Development Manager
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6
Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com
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From juanito1982 at gmail.com  Mon Jun  6 11:01:04 2011
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Mon, 6 Jun 2011 09:01:04 +0200
Subject: [Freeswitch-users] Fwd: Major deployment of outbound FAX on
 latest version of Freeswitch question
In-Reply-To: <4DCBE7D4.1070100@coppice.org>
References: <C42A22554C0CF44DAC0AA1F132B86DE65819F71BAC@ASKSERVER.asksbs.local>
	<4DC40C86.9000503@coppice.org>
	<C42A22554C0CF44DAC0AA1F132B86DE658284AF104@ASKSERVER.asksbs.local>
	<4DC67898.2090106@coppice.org>
	<BANLkTin7OXHYE2V5d3AnWeNJAHhYQeJk9g@mail.gmail.com>
	<4DC732BC.9070206@coppice.org>
	<BANLkTiktkLdRSHWZFVv9JGz5hzP7cAay=w@mail.gmail.com>
	<4DCBE7D4.1070100@coppice.org>
Message-ID: <BANLkTik8s+ufq3TsVUUxV2nPsRmkW5vwTg@mail.gmail.com>

Having FS plus a Sangoma Card with echo cancel, is necessary to disable echo
cancel in order to sending/receiving faxes? How can it be done?

Regards

2011/5/12 Steve Underwood <steveu at coppice.org>

> On 05/12/2011 05:21 PM, Juan Antonio Iba?ez Santorum wrote:
> > Hello Steve!
> >
> >    I would like to ask you one question about fax system. You told
> > that using FS+mod_spandsp could be gotten successful faxing rates over
> > 99%. Looking at your experience, is 99% nearer to 100% or to 99%?
> > Which errors do you find usually on that 1%?
> >
> You can run tests all day in a lab without failure. The real test is how
> well things work against a wide diversity of machines in the real world.
> This http://www.soft-switch.org/spandsp-soft-fax-performance.html says
> something about our testing, and what is achievable. Its quite a lot of
> work to get real reliability numbers. You get a lot of call failures on
> a typical public FAX server due to things like wrong numbers, voice
> calls into a FAX port, people purposefully dropping half completed FAX
> calls, and buggy FAX machines doing weird things. Don't ignore the last
> one. There are famous make machines with serious bugs. You need to
> manually check the failed calls one by one to get real reliability
> figures. In the end there are always a few calls where you are never
> sure who's fault the failure is. All I can say with hand on heart is the
> number of unexplainable failures is well below 1%
>
> Regards,
> Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From david.villasmil.work at gmail.com  Mon Jun  6 16:15:39 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 6 Jun 2011 14:15:39 +0200
Subject: [Freeswitch-users] XML_CURL and configuration missing
Message-ID: <BANLkTimfNtVo49ebqqkqi2BVj2+e7_ANsw@mail.gmail.com>

Hello all,

I have xml_curl to load sofia and distributor, it's working fine... but
after some calls (don't know how many) FS starts giving:

2011-06-06 13:33:10.913489 [ERR] mod_sofia.c:4036 Invalid Gateway

And calls are not going through.

When this happens I do:

 sofia status

and I get: (???)

sofia status
                     Name          Type
  Data      State
=================================================================================================
                     gold       my_profile
sip:mod_sofia at MY_LOCAL_IP:5060      RUNNING (0)
=================================================================================================

then i RESTART the profile:

sofia profile restart all

and again:

sofia status

and i get:

                     Name          Type
  Data      State
=================================================================================================
                     my_profile       profile
sip:mod_sofia at MY_LOCAL_IP:5060      RUNNING (0)
            my_profile::GW1       gateway                sip:1234 at GW1_IP:5060
     NOREG
             my_profile::GW2       gateway                sip:1234 at GW2_IP:5066
     NOREG
            my_profile::GW3       gateway             sip:1111111 at GW3_IP:5060
     NOREG
               my_profile::GW4       gateway
sip:1234 at GW4_IP:5060
     NOREG
=================================================================================================
1 profile 0 aliases



the obvious question is: Why is this happening???

thanks all for your help!

David
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From steveayre at gmail.com  Mon Jun  6 17:39:08 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 6 Jun 2011 14:39:08 +0100
Subject: [Freeswitch-users] XML_CURL and configuration missing
In-Reply-To: <BANLkTimfNtVo49ebqqkqi2BVj2+e7_ANsw@mail.gmail.com>
References: <BANLkTimfNtVo49ebqqkqi2BVj2+e7_ANsw@mail.gmail.com>
Message-ID: <BANLkTinsmoUGx1wAWowW9gLB7x+R3TOSSA@mail.gmail.com>

Does this happen on the latest git?

Are there any other messages prior to the ERR that might indicate when/why
the gateways disappear?

-Steve



On 6 June 2011 13:15, David Villasmil <david.villasmil.work at gmail.com>wrote:

> Hello all,
>
> I have xml_curl to load sofia and distributor, it's working fine... but
> after some calls (don't know how many) FS starts giving:
>
> 2011-06-06 13:33:10.913489 [ERR] mod_sofia.c:4036 Invalid Gateway
>
> And calls are not going through.
>
> When this happens I do:
>
>  sofia status
>
> and I get: (???)
>
> sofia status
>                      Name          Type
>   Data      State
>
> =================================================================================================
>                      gold       my_profile
> sip:mod_sofia at MY_LOCAL_IP:5060      RUNNING (0)
>
> =================================================================================================
>
> then i RESTART the profile:
>
> sofia profile restart all
>
> and again:
>
> sofia status
>
> and i get:
>
>                      Name          Type
>   Data      State
>
> =================================================================================================
>                      my_profile       profile
> sip:mod_sofia at MY_LOCAL_IP:5060      RUNNING (0)
>             my_profile::GW1       gateway                sip:1234 at GW1_IP:5060
>      NOREG
>              my_profile::GW2       gateway                sip:1234 at GW2_IP:5066
>      NOREG
>             my_profile::GW3       gateway             sip:1111111 at GW3_IP:5060
>      NOREG
>                my_profile::GW4       gateway
> sip:1234 at GW4_IP:5060      NOREG
>
> =================================================================================================
> 1 profile 0 aliases
>
>
>
> the obvious question is: Why is this happening???
>
> thanks all for your help!
>
> David
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.villasmil.work at gmail.com  Mon Jun  6 17:50:46 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 6 Jun 2011 15:50:46 +0200
Subject: [Freeswitch-users] XML_CURL and configuration missing
In-Reply-To: <BANLkTinsmoUGx1wAWowW9gLB7x+R3TOSSA@mail.gmail.com>
References: <BANLkTimfNtVo49ebqqkqi2BVj2+e7_ANsw@mail.gmail.com>
	<BANLkTinsmoUGx1wAWowW9gLB7x+R3TOSSA@mail.gmail.com>
Message-ID: <BANLkTi=YSYG5-_yisshGiEOnAB63gvsztA@mail.gmail.com>

Thanks for answering Steve.

No, i don't see any errors anywhere...
I haven't tried the latest git, and i'm reluctant to update as I have an app
running on this server... don't want any backward-compatibility issues... if
this is a known problem, then i will have to update, else I would rather try
and find what the problem is before updating.


my version is:

FreeSWITCH Version 1.0.head (git-7e52acf 2011-03-28 22-18-47 -0500)

Thanks!


On Mon, Jun 6, 2011 at 3:39 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Does this happen on the latest git?
>
> Are there any other messages prior to the ERR that might indicate when/why
> the gateways disappear?
>
> -Steve
>
>
>
> On 6 June 2011 13:15, David Villasmil <david.villasmil.work at gmail.com>wrote:
>
>>  Hello all,
>>
>> I have xml_curl to load sofia and distributor, it's working fine... but
>> after some calls (don't know how many) FS starts giving:
>>
>> 2011-06-06 13:33:10.913489 [ERR] mod_sofia.c:4036 Invalid Gateway
>>
>> And calls are not going through.
>>
>> When this happens I do:
>>
>>  sofia status
>>
>> and I get: (???)
>>
>> sofia status
>>                      Name          Type
>>     Data      State
>>
>> =================================================================================================
>>                      gold       my_profile
>> sip:mod_sofia at MY_LOCAL_IP:5060      RUNNING (0)
>>
>> =================================================================================================
>>
>> then i RESTART the profile:
>>
>> sofia profile restart all
>>
>> and again:
>>
>> sofia status
>>
>> and i get:
>>
>>                      Name          Type
>>     Data      State
>>
>> =================================================================================================
>>                      my_profile       profile
>> sip:mod_sofia at MY_LOCAL_IP:5060      RUNNING (0)
>>             my_profile::GW1       gateway                sip:1234 at GW1_IP:5060
>>      NOREG
>>              my_profile::GW2       gateway                sip:1234 at GW2_IP:5066
>>      NOREG
>>             my_profile::GW3       gateway             sip:1111111 at GW3_IP:5060
>>      NOREG
>>                my_profile::GW4       gateway
>> sip:1234 at GW4_IP:5060      NOREG
>>
>> =================================================================================================
>> 1 profile 0 aliases
>>
>>
>>
>> the obvious question is: Why is this happening???
>>
>> thanks all for your help!
>>
>> David
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Mon Jun  6 18:13:33 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 6 Jun 2011 15:13:33 +0100
Subject: [Freeswitch-users] XML_CURL and configuration missing
In-Reply-To: <BANLkTi=YSYG5-_yisshGiEOnAB63gvsztA@mail.gmail.com>
References: <BANLkTimfNtVo49ebqqkqi2BVj2+e7_ANsw@mail.gmail.com>
	<BANLkTinsmoUGx1wAWowW9gLB7x+R3TOSSA@mail.gmail.com>
	<BANLkTi=YSYG5-_yisshGiEOnAB63gvsztA@mail.gmail.com>
Message-ID: <BANLkTi=436Dmz0TxN+YkPci4rvi8kZDEzA@mail.gmail.com>

I think I do remember seeing this reported once before, which might mean
that it was looked into and fixed. I can't say for sure though.

It might be worth looking at the commit log on fisheye and bug reports on
jira to see if there's anything that sounds like this mentioned.

-Steve



On 6 June 2011 14:50, David Villasmil <david.villasmil.work at gmail.com>wrote:

> Thanks for answering Steve.
>
> No, i don't see any errors anywhere...
> I haven't tried the latest git, and i'm reluctant to update as I have an
> app running on this server... don't want any backward-compatibility
> issues... if this is a known problem, then i will have to update, else I
> would rather try and find what the problem is before updating.
>
>
> my version is:
>
> FreeSWITCH Version 1.0.head (git-7e52acf 2011-03-28 22-18-47 -0500)
>
> Thanks!
>
>
> On Mon, Jun 6, 2011 at 3:39 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Does this happen on the latest git?
>>
>> Are there any other messages prior to the ERR that might indicate when/why
>> the gateways disappear?
>>
>> -Steve
>>
>>
>>
>> On 6 June 2011 13:15, David Villasmil <david.villasmil.work at gmail.com>wrote:
>>
>>>  Hello all,
>>>
>>> I have xml_curl to load sofia and distributor, it's working fine... but
>>> after some calls (don't know how many) FS starts giving:
>>>
>>> 2011-06-06 13:33:10.913489 [ERR] mod_sofia.c:4036 Invalid Gateway
>>>
>>> And calls are not going through.
>>>
>>> When this happens I do:
>>>
>>>  sofia status
>>>
>>> and I get: (???)
>>>
>>> sofia status
>>>                      Name          Type
>>>     Data      State
>>>
>>> =================================================================================================
>>>                      gold       my_profile
>>> sip:mod_sofia at MY_LOCAL_IP:5060      RUNNING (0)
>>>
>>> =================================================================================================
>>>
>>> then i RESTART the profile:
>>>
>>> sofia profile restart all
>>>
>>> and again:
>>>
>>> sofia status
>>>
>>> and i get:
>>>
>>>                      Name          Type
>>>     Data      State
>>>
>>> =================================================================================================
>>>                      my_profile       profile
>>> sip:mod_sofia at MY_LOCAL_IP:5060      RUNNING (0)
>>>             my_profile::GW1       gateway                sip:1234 at GW1_IP:5060
>>>      NOREG
>>>              my_profile::GW2       gateway                sip:1234 at GW2_IP:5066
>>>      NOREG
>>>             my_profile::GW3       gateway             sip:1111111 at GW3_IP:5060
>>>      NOREG
>>>                my_profile::GW4       gateway
>>> sip:1234 at GW4_IP:5060      NOREG
>>>
>>> =================================================================================================
>>> 1 profile 0 aliases
>>>
>>>
>>>
>>> the obvious question is: Why is this happening???
>>>
>>> thanks all for your help!
>>>
>>> David
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From adam.kelloway at newpace.ca  Mon Jun  6 18:25:19 2011
From: adam.kelloway at newpace.ca (Adam Kelloway)
Date: Mon, 06 Jun 2011 11:25:19 -0300
Subject: [Freeswitch-users] proposed patch to org.freeswitch.esl.client
Message-ID: <4DECE34F.2090909@newpace.ca>

Hi there,

This message is to the author of org.freeswitch.esl.client:

I have attached a git diff of some small changes I was hoping could be
made to the org.freeswitch.esl.client library. To summarize:

1. Added EslMessage parameter to handleDisconnectionNotice
2. Added some more headers to EslHeaders

My motivation for this is that I wanted be able to read the headers in a
disconnection EslMessage. The first problem was that
handleDisconnectionNotice didn't pass in the EslMessage. The second
problem was that the Name object didn't define the headers I wanted,
resulting in them being set in the headers map with a key of null.

Do you think this is a reasonable change to your code? If so, can it be
included in the next release? On that note, when do you think the next
release will be made?

Thanks and keep up the great work,

Adam


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From jeff at jefflenk.com  Mon Jun  6 18:25:41 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Mon, 6 Jun 2011 07:25:41 -0700 (PDT)
Subject: [Freeswitch-users] Click-to-call from Outlook
In-Reply-To: <SNT116-W414898E6E31CC59968CBA79D7F0@phx.gbl>
References: <SNT116-W414898E6E31CC59968CBA79D7F0@phx.gbl>
Message-ID: <1307370341109-6445362.post@n2.nabble.com>

checkout siptapi the last time i tried it it worked well for fs

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Click-to-call-from-Outlook-tp6435767p6445362.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From mail.mthakkar at gmail.com  Mon Jun  6 14:32:59 2011
From: mail.mthakkar at gmail.com (Mitesh Thakkar)
Date: Mon, 6 Jun 2011 16:02:59 +0530
Subject: [Freeswitch-users]  cancel a non answered bridge
In-Reply-To: <BANLkTi=bXjF=ePMHutbj5UxXtCF-GX69Sw@mail.gmail.com>
References: <BANLkTi=bXjF=ePMHutbj5UxXtCF-GX69Sw@mail.gmail.com>
Message-ID: <BANLkTikvK1Q_u3Y6xSuoNAgO0xHgzu_B-g@mail.gmail.com>

Hello all,

I am facing the similar kind of issue. My flow is as below.
1. doing bridge to route 1.
2. on fail i need some work to be done.
3. doing bridge to route 2.

So, in above (1.) is being timeout with progress_timeout., and process
further and does bridge to route 2. It sends INVITE to route 2 but it is not
CANCELing to route 1 and keeps sending INVITEs.

Basically, it should not send INVITE once skipped. Kindly guide me how can I
achieve it.

Regards,
Mitesh Thakkar


> the situation :
>
> Aleg call external gateway
> in dialplan I set on the fly the nibble_rate (nibble_account is already set)
> and just after make a bridge to the gateway.
> so nibblebill tries to ask DB if cahs is ok (so it takes a little time) but in the mean
> time the bridge is created and Bleg is ringing.
> nibblebill realize that cash is below the amount allowed and redirect to nofunds extension.
>
> I'd like to avoid the Bleg ringing by kill the bridge (but not hanup Aleg) once the nofunds extension is reached.
>
> Thanks
>   ----- Original Message -----
>   From: David Ponzone
>   To: FreeSWITCH Users Help
>   Sent: Friday, January 28, 2011 12:27 PM
>   Subject: Re: [Freeswitch-users] cancel a non answered bridge
>
>
>   Sorry, I fail to get the question...
>   Perhaps we need more details like: where do you want to do that ? from XML dialplan ? from script ?
>   From API, you can uuid_kill.
>   From XML, I am not sure I see what that would mean...perhaps you need to define a timeout, so that the bridge attempt ends after X seconds ?
>
>
>   David Ponzone  Direction Technique
>   email: david.ponzone at ipeva.fr <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>   tel:      01 74 03 18 97
>   gsm:   06 66 98 76 34
>
>
>   Service Client IPeva
>   tel:      0811 46 26 26
>   www.ipeva.fr  -   www.ipeva-studio.com
>
>
>   Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
>
>
>
>   Le 28/01/2011 ? 18:14, Madovsky a ?crit :
>
>
>     Sorry I forgot
>
>     without to hangup ?
>       ----- Original Message -----
>       From: Madovsky
>       To: freeswitch-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>       Sent: Friday, January 28, 2011 12:13 PM
>       Subject: cancel a non answered bridge
>
>
>       How to cancel ok kil a non answered bridge ?
>
>       Thanks
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
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From matzemuc86 at gmail.com  Mon Jun  6 17:40:49 2011
From: matzemuc86 at gmail.com (Matthias Kaufmann)
Date: Mon, 6 Jun 2011 15:40:49 +0200
Subject: [Freeswitch-users] Compile FreeSWITCH on WinXP, VS2008 pro
Message-ID: <BANLkTinj__wxkcmu3rfbsSwoPnfNDhRiYQ@mail.gmail.com>

Hello,

I try to compile FreeSWITCh with VS2008 pro Version on my WinXP (32bit) OS
but I get 4 errors. Where can I find the log to get more information?
Starting freewitch works but e.g. mod_sofia is not working :-(.

Thanks
MatzeMuc86
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From steveayre at gmail.com  Mon Jun  6 18:58:52 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 6 Jun 2011 15:58:52 +0100
Subject: [Freeswitch-users] Compile FreeSWITCH on WinXP, VS2008 pro
In-Reply-To: <BANLkTinj__wxkcmu3rfbsSwoPnfNDhRiYQ@mail.gmail.com>
References: <BANLkTinj__wxkcmu3rfbsSwoPnfNDhRiYQ@mail.gmail.com>
Message-ID: <BANLkTik83p5i2do8chwb9pSELQeDkWYKRg@mail.gmail.com>

Can you tell us what the errors are? Are they compile errors, or runtime
errors?

-Steve


On 6 June 2011 14:40, Matthias Kaufmann <matzemuc86 at gmail.com> wrote:

> Hello,
>
> I try to compile FreeSWITCh with VS2008 pro Version on my WinXP (32bit) OS
> but I get 4 errors. Where can I find the log to get more information?
> Starting freewitch works but e.g. mod_sofia is not working :-(.
>
> Thanks
> MatzeMuc86
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From matzemuc86 at gmail.com  Mon Jun  6 19:02:01 2011
From: matzemuc86 at gmail.com (Matthias Kaufmann)
Date: Mon, 6 Jun 2011 17:02:01 +0200
Subject: [Freeswitch-users] Compile FreeSWITCH on WinXP, VS2008 pro
In-Reply-To: <BANLkTik83p5i2do8chwb9pSELQeDkWYKRg@mail.gmail.com>
References: <BANLkTinj__wxkcmu3rfbsSwoPnfNDhRiYQ@mail.gmail.com>
	<BANLkTik83p5i2do8chwb9pSELQeDkWYKRg@mail.gmail.com>
Message-ID: <BANLkTim8fBKJcNJXtLjVD4vr6T+AXjmFXQ@mail.gmail.com>

compile errors in VS2008.

Viele Gr??e
Matthias Kaufmann



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

> Can you tell us what the errors are? Are they compile errors, or runtime
> errors?
>
> -Steve
>
>
> On 6 June 2011 14:40, Matthias Kaufmann <matzemuc86 at gmail.com> wrote:
>
>> Hello,
>>
>> I try to compile FreeSWITCh with VS2008 pro Version on my WinXP (32bit) OS
>> but I get 4 errors. Where can I find the log to get more information?
>> Starting freewitch works but e.g. mod_sofia is not working :-(.
>>
>> Thanks
>> MatzeMuc86
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jeff at jefflenk.com  Mon Jun  6 19:35:04 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Mon, 6 Jun 2011 08:35:04 -0700 (PDT)
Subject: [Freeswitch-users] Compile FreeSWITCH on WinXP, VS2008 pro
In-Reply-To: <BANLkTim8fBKJcNJXtLjVD4vr6T+AXjmFXQ@mail.gmail.com>
References: <BANLkTinj__wxkcmu3rfbsSwoPnfNDhRiYQ@mail.gmail.com>
	<BANLkTik83p5i2do8chwb9pSELQeDkWYKRg@mail.gmail.com>
	<BANLkTim8fBKJcNJXtLjVD4vr6T+AXjmFXQ@mail.gmail.com>
Message-ID: <1307374504805-6445607.post@n2.nabble.com>

Just grab the output window in visual studio it has the full build log and
attach a pastebin here

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Compile-FreeSWITCH-on-WinXP-VS2008-pro-tp6445427p6445607.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Mon Jun  6 19:56:37 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 6 Jun 2011 16:56:37 +0100
Subject: [Freeswitch-users] Compile FreeSWITCH on WinXP, VS2008 pro
In-Reply-To: <1307374504805-6445607.post@n2.nabble.com>
References: <BANLkTinj__wxkcmu3rfbsSwoPnfNDhRiYQ@mail.gmail.com>
	<BANLkTik83p5i2do8chwb9pSELQeDkWYKRg@mail.gmail.com>
	<BANLkTim8fBKJcNJXtLjVD4vr6T+AXjmFXQ@mail.gmail.com>
	<1307374504805-6445607.post@n2.nabble.com>
Message-ID: <BANLkTim6_7-TR4kEW29JECkxGYnd_o_Tfw@mail.gmail.com>

Indeed. Also check you're using Freeswitch.2008.sln, and that you're using
the latest version of FS from Git.

-Steve


On 6 June 2011 16:35, Jeff Lenk <jeff at jefflenk.com> wrote:

> Just grab the output window in visual studio it has the full build log and
> attach a pastebin here
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Compile-FreeSWITCH-on-WinXP-VS2008-pro-tp6445427p6445607.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From a.afzali2003 at gmail.com  Mon Jun  6 20:18:34 2011
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Mon, 6 Jun 2011 20:48:34 +0430
Subject: [Freeswitch-users] Binding meta apps on mod_fifo dispatched calls
Message-ID: <BANLkTinshubu8rHshfBTzmiQc4U-+NOSKA@mail.gmail.com>

Hi Guys,

It seems mod_fifo enables *dx* feature extension ( * 1 ) on all calls which
bridge to agents by default. I want to be able to add *cf* feature extension
as well. Is there a way except that to use mod_loopback?

BEST,
-- afshin
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From yungwei at resolvity.com  Mon Jun  6 21:51:01 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Mon, 6 Jun 2011 13:51:01 -0400
Subject: [Freeswitch-users] Does FreeSwitch support something like "preserve
 bandwidth during silent periods"?
Message-ID: <33095823FD21DF429B481B5163264B7950AC585FF9@VMBX102.ihostexchange.net>

Hi,

I'm just wondering if FreeSwitch allows us to control whether to preserve bandwidth during silent periods in config files. Thanks.


From steveayre at gmail.com  Mon Jun  6 22:04:00 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 6 Jun 2011 19:04:00 +0100
Subject: [Freeswitch-users] Does FreeSwitch support something like
 "preserve bandwidth during silent periods"?
In-Reply-To: <33095823FD21DF429B481B5163264B7950AC585FF9@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950AC585FF9@VMBX102.ihostexchange.net>
Message-ID: <BANLkTi=pFLgKcPDG24g2S-2E6t_ydbPewQ@mail.gmail.com>

Yes:

http://wiki.freeswitch.org/wiki/VAD_and_CNG

-Steve



On 6 June 2011 18:51, Yungwei Chen <yungwei at resolvity.com> wrote:

> Hi,
>
> I'm just wondering if FreeSwitch allows us to control whether to preserve
> bandwidth during silent periods in config files. Thanks.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From freeswitch at aastral.net  Mon Jun  6 22:24:18 2011
From: freeswitch at aastral.net (Bill W.)
Date: Mon, 06 Jun 2011 14:24:18 -0400
Subject: [Freeswitch-users] intermittent EXCHANGE_ROUTING_ERROR
In-Reply-To: <BANLkTink0R3qq96hQez786NJhFM_yAzrhg@mail.gmail.com>
References: <1QRrkF-0007cj-IK@mail.aastral.net>	<BANLkTikc5bMokDY6__miju=rOUmsLc7kNg@mail.gmail.com>	<1QRt7M-0002Zm-GE@mail.aastral.net>	<BANLkTikXJrxuwUXAd2kYFCiDQOLGQgw9dg@mail.gmail.com>	<1QS7yM-0005cW-8h@mail.aastral.net>	<BANLkTi=hv5KZhMUfj1TKbucARH8gzADqpw@mail.gmail.com>	<1QSAQM-0003GJ-6U@mail.aastral.net>
	<BANLkTink0R3qq96hQez786NJhFM_yAzrhg@mail.gmail.com>
Message-ID: <1QTeTP-0003ff-Om@mail.aastral.net>

Hey Anthony/Michael,

After a lot of digging, it turns out the problem was the Max-Forwards on
the incoming SIP traffic was too low for FreeSWITCH to complete the call.

Thanks so much for all your help!




On 6/2/11 1:13 PM, Anthony Minessale wrote:
> then look for a low max-forwards header in the sip traffic
> 
> On Thu, Jun 2, 2011 at 11:07 AM, Bill W. <freeswitch at aastral.net> wrote:
>>
>> Hey Anthony,
>>
>> All the bridge commands are of the form:
>> sofia/profile/18005551212 at 1.2.3.4
>>
>> Also, if it was looping, would I see that in the logs?  It would parse
>> the dialplan 73 times, correct?  Because I don't see that.
>>


From fborot at hotmail.com  Mon Jun  6 23:16:25 2011
From: fborot at hotmail.com (Fabian Borot)
Date: Mon, 6 Jun 2011 15:16:25 -0400
Subject: [Freeswitch-users] Questions
Message-ID: <SNT110-W43983F1D6DACAB71BE8CE1CD600@phx.gbl>


Hello

We are giving Freeswitch a try in our organization to provide H323 <-> SIP functionality. We have 2 main questions/requests before starting installing/configuring:

1- What would be the most recommended module for H323? I see there are two [opal and mod_h323] both both say that are in development stages. 

2- Does FS fully supports virtual interfaces [eth0:1]? I mean: 
        2- 1 listening on several virtual interfaces,
        2-2 replying from the virtual interface that the request came in instead of replying from the primary interface
        2-3 and also we need to specify the interface to send the call out instead of using the primary interface

Thank you

Fabian





 

 		 	   		  
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From msc at freeswitch.org  Tue Jun  7 01:07:01 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 6 Jun 2011 14:07:01 -0700
Subject: [Freeswitch-users] intermittent EXCHANGE_ROUTING_ERROR
In-Reply-To: <1QTeTP-0003ff-Om@mail.aastral.net>
References: <1QRrkF-0007cj-IK@mail.aastral.net>
	<BANLkTikc5bMokDY6__miju=rOUmsLc7kNg@mail.gmail.com>
	<1QRt7M-0002Zm-GE@mail.aastral.net>
	<BANLkTikXJrxuwUXAd2kYFCiDQOLGQgw9dg@mail.gmail.com>
	<1QS7yM-0005cW-8h@mail.aastral.net>
	<BANLkTi=hv5KZhMUfj1TKbucARH8gzADqpw@mail.gmail.com>
	<1QSAQM-0003GJ-6U@mail.aastral.net>
	<BANLkTink0R3qq96hQez786NJhFM_yAzrhg@mail.gmail.com>
	<1QTeTP-0003ff-Om@mail.aastral.net>
Message-ID: <BANLkTiko++ShiZKb_+bDNPprirrLeXp2RQ@mail.gmail.com>

Thanks for the followup email. It's always nice to know how these problems
eventually get solved...

-MC

On Mon, Jun 6, 2011 at 11:24 AM, Bill W. <freeswitch at aastral.net> wrote:

> Hey Anthony/Michael,
>
> After a lot of digging, it turns out the problem was the Max-Forwards on
> the incoming SIP traffic was too low for FreeSWITCH to complete the call.
>
> Thanks so much for all your help!
>
>
>
>
> On 6/2/11 1:13 PM, Anthony Minessale wrote:
> > then look for a low max-forwards header in the sip traffic
> >
> > On Thu, Jun 2, 2011 at 11:07 AM, Bill W. <freeswitch at aastral.net> wrote:
> >>
> >> Hey Anthony,
> >>
> >> All the bridge commands are of the form:
> >> sofia/profile/18005551212 at 1.2.3.4
> >>
> >> Also, if it was looping, would I see that in the logs?  It would parse
> >> the dialplan 73 times, correct?  Because I don't see that.
> >>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kahn at vestec.com  Tue Jun  7 01:08:23 2011
From: kahn at vestec.com (Kashif Kahn)
Date: Mon, 06 Jun 2011 17:08:23 -0400
Subject: [Freeswitch-users] Speech Recognition: Indian English
Message-ID: <4DED41C7.4050009@vestec.com>

Hello Everyone,

Vestec ASR engine now supports speech recognition in Indian English. (We 
also support American English, Australian English, and Chinese 
Mandarin). A starter kit is available for $25: 
http://www.vestec.com/products

Please note that Vestec offers the best deal around for enabling 
sophisticated speech recognition with command-and-control type IVR 
applications by offering a high accuracy, standards based speech engine 
at a fraction of the cost of conventional ASR vendors. Feel free to 
contact me with any questions or concerns.

Regards,
-Kashif



From msc at freeswitch.org  Tue Jun  7 01:10:32 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 6 Jun 2011 14:10:32 -0700
Subject: [Freeswitch-users] cancel a non answered bridge
In-Reply-To: <BANLkTikvK1Q_u3Y6xSuoNAgO0xHgzu_B-g@mail.gmail.com>
References: <BANLkTi=bXjF=ePMHutbj5UxXtCF-GX69Sw@mail.gmail.com>
	<BANLkTikvK1Q_u3Y6xSuoNAgO0xHgzu_B-g@mail.gmail.com>
Message-ID: <BANLkTint0rCZGrmRCGYYLganDaiFdLTHmA@mail.gmail.com>

Can you supply a debug log w/ siptrace? Also include the dialplan that you
are using.
-MC

On Mon, Jun 6, 2011 at 3:32 AM, Mitesh Thakkar <mail.mthakkar at gmail.com>wrote:

> Hello all,
>
> I am facing the similar kind of issue. My flow is as below.
> 1. doing bridge to route 1.
> 2. on fail i need some work to be done.
> 3. doing bridge to route 2.
>
> So, in above (1.) is being timeout with progress_timeout., and process
> further and does bridge to route 2. It sends INVITE to route 2 but it is not
> CANCELing to route 1 and keeps sending INVITEs.
>
> Basically, it should not send INVITE once skipped. Kindly guide me how can
> I achieve it.
>
> Regards,
> Mitesh Thakkar
>
>
>> the situation :
>>
>> Aleg call external gateway
>> in dialplan I set on the fly the nibble_rate (nibble_account is already set)
>> and just after make a bridge to the gateway.
>> so nibblebill tries to ask DB if cahs is ok (so it takes a little time) but in the mean
>> time the bridge is created and Bleg is ringing.
>> nibblebill realize that cash is below the amount allowed and redirect to nofunds extension.
>>
>> I'd like to avoid the Bleg ringing by kill the bridge (but not hanup Aleg) once the nofunds extension is reached.
>>
>> Thanks
>>   ----- Original Message -----
>>   From: David Ponzone
>>   To: FreeSWITCH Users Help
>>   Sent: Friday, January 28, 2011 12:27 PM
>>   Subject: Re: [Freeswitch-users] cancel a non answered bridge
>>
>>
>>   Sorry, I fail to get the question...
>>   Perhaps we need more details like: where do you want to do that ? from XML dialplan ? from script ?
>>   From API, you can uuid_kill.
>>   From XML, I am not sure I see what that would mean...perhaps you need to define a timeout, so that the bridge attempt ends after X seconds ?
>>
>>
>>   David Ponzone  Direction Technique
>>   email: david.ponzone at ipeva.fr <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>   tel:      01 74 03 18 97
>>   gsm:   06 66 98 76 34
>>
>>
>>   Service Client IPeva
>>   tel:      0811 46 26 26
>>   www.ipeva.fr  -   www.ipeva-studio.com
>>
>>
>>   Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>>
>>
>>
>>
>>   Le 28/01/2011 ? 18:14, Madovsky a ?crit :
>>
>>
>>     Sorry I forgot
>>
>>     without to hangup ?
>>       ----- Original Message -----
>>       From: Madovsky
>>       To: freeswitch-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>       Sent: Friday, January 28, 2011 12:13 PM
>>       Subject: cancel a non answered bridge
>>
>>
>>       How to cancel ok kil a non answered bridge ?
>>
>>       Thanks
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chat2jesse at gmail.com  Tue Jun  7 03:10:20 2011
From: chat2jesse at gmail.com (jesse)
Date: Mon, 6 Jun 2011 16:10:20 -0700
Subject: [Freeswitch-users] Freeswitch 1.0.6 doesn't have mod_hash.c?
Message-ID: <BANLkTik5eLVYmcQfcDBjD6natOdtsVVLDFD5jSxyCWAybG6-SQ@mail.gmail.com>

but my fs running time complains : switch_core_session.c:1731 Invalid
Application hash. I couldn't find mod_hash.c in src package...

any idea?

thanks!
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From michal.bielicki at seventhsignal.de  Tue Jun  7 03:13:05 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Tue, 07 Jun 2011 01:13:05 +0200
Subject: [Freeswitch-users] Does FreeSwitch support something like
 "preserve bandwidth during silent periods"?
In-Reply-To: <33095823FD21DF429B481B5163264B7950AC585FF9@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950AC585FF9@VMBX102.ihostexchange.net>
Message-ID: <4DED5F01.3000600@seventhsignal.de>

Am 06.06.2011 19:51, schrieb Yungwei Chen:
> Hi,
>
> I'm just wondering if FreeSwitch allows us to control whether to preserve bandwidth during silent periods in config files. Thanks.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
yes


From frankie.k.yiu at gmail.com  Tue Jun  7 03:29:35 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Mon, 6 Jun 2011 16:29:35 -0700
Subject: [Freeswitch-users] How to stop a Recording when the user presses
	"#" on C# project?
Message-ID: <BANLkTikRvoKtJ=a+96_jWa5S=WSOp8sc2g@mail.gmail.com>

Hi there,

I am working on a C# project and would like to record a message when the
user is ready to record.  Here is the API for C#:
* public int RecordFile(string file_name, int time_limit, int
silence_threshold, int silence_hits);

*This API will stop the call only on timeout or when there is seconds of
silence.  What I want to do is to let the user decide when to end the
recording by pressing a "#" key.  So how can I do that?  Do I have to use C
code to do that? (by sending event to C code side?)

Thanks,
Frankie
*

*
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From bobc at devassert.com  Tue Jun  7 05:31:10 2011
From: bobc at devassert.com (Bob Coleman)
Date: Tue, 7 Jun 2011 13:31:10 +1200
Subject: [Freeswitch-users] How to stop a Recording when the user
 presses "#" on C# project?
In-Reply-To: <BANLkTikRvoKtJ=a+96_jWa5S=WSOp8sc2g@mail.gmail.com>
References: <BANLkTikRvoKtJ=a+96_jWa5S=WSOp8sc2g@mail.gmail.com>
Message-ID: <BANLkTikqx5hyDa39mrTQEFGuSiTHNy7t-g@mail.gmail.com>

Hi,

You just set the playback_terminators to a # before you issue the record

Cheers

Bob

On Tue, Jun 7, 2011 at 11:29 AM, Frankie Yiu <frankie.k.yiu at gmail.com> wrote:
> Hi there,
>
> I am working on a C# project and would like to record a message when the
> user is ready to record.? Here is the API for C#:
> ?public int RecordFile(string file_name, int time_limit, int
> silence_threshold, int silence_hits);
>
> This API will stop the call only on timeout or when there is seconds of
> silence.? What I want to do is to let the user decide when to end the
> recording by pressing a "#" key.? So how can I do that?? Do I have to use C
> code to do that? (by sending event to C code side?)
>
> Thanks,
> Frankie
>


From boris at tagnet.ru  Tue Jun  7 07:47:28 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Tue, 07 Jun 2011 09:47:28 +0600
Subject: [Freeswitch-users] Freeswitch 1.0.6 doesn't have mod_hash.c?
In-Reply-To: <BANLkTik5eLVYmcQfcDBjD6natOdtsVVLDFD5jSxyCWAybG6-SQ@mail.gmail.com>
References: <BANLkTik5eLVYmcQfcDBjD6natOdtsVVLDFD5jSxyCWAybG6-SQ@mail.gmail.com>
Message-ID: <4DED9F50.50109@tagnet.ru>

Hello!

     It is in source tree now. Please read Changelog

> but my fs running time complains : switch_core_session.c:1731 Invalid 
> Application hash. I couldn't find mod_hash.c in src package...
>
> any idea?
>
> thanks!
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Regards,
Boris


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From adminjew at gmail.com  Tue Jun  7 07:55:51 2011
From: adminjew at gmail.com (Yitzchok)
Date: Mon, 6 Jun 2011 23:55:51 -0400
Subject: [Freeswitch-users] How to stop a Recording when the user
 presses "#" on C# project?
In-Reply-To: <BANLkTikqx5hyDa39mrTQEFGuSiTHNy7t-g@mail.gmail.com>
References: <BANLkTikRvoKtJ=a+96_jWa5S=WSOp8sc2g@mail.gmail.com>
	<BANLkTikqx5hyDa39mrTQEFGuSiTHNy7t-g@mail.gmail.com>
Message-ID: <BANLkTimA5AsADJkfsKqtRhC55fMN8T=PdA@mail.gmail.com>

session.SetVariable("playback_terminators", "#");
session.RecordFile(fileName, 7000, 1000, 5);

This should also work if you need something more advanced.

var receivedFunction = (c, t) => (c == '#') ? "break" : "";
session.DtmfReceivedFunction += receivedFunction;
session.RecordFile(fileName, 7000, 1000, 5);
session.DtmfReceivedFunction -= receivedFunction;


Yitzchok


On Mon, Jun 6, 2011 at 9:31 PM, Bob Coleman <bobc at devassert.com> wrote:

> Hi,
>
> You just set the playback_terminators to a # before you issue the record
>
> Cheers
>
> Bob
>
> On Tue, Jun 7, 2011 at 11:29 AM, Frankie Yiu <frankie.k.yiu at gmail.com>
> wrote:
> > Hi there,
> >
> > I am working on a C# project and would like to record a message when the
> > user is ready to record.  Here is the API for C#:
> >  public int RecordFile(string file_name, int time_limit, int
> > silence_threshold, int silence_hits);
> >
> > This API will stop the call only on timeout or when there is seconds of
> > silence.  What I want to do is to let the user decide when to end the
> > recording by pressing a "#" key.  So how can I do that?  Do I have to use
> C
> > code to do that? (by sending event to C code side?)
> >
> > Thanks,
> > Frankie
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From freeswitch at earthspike.net  Tue Jun  7 01:36:20 2011
From: freeswitch at earthspike.net (John)
Date: Mon, 06 Jun 2011 22:36:20 +0100
Subject: [Freeswitch-users] FreeTDM - Sangoma B700 - ISDN connection
	questions - UK
Message-ID: <4DED4854.6010203@earthspike.net>

Hello,

I have just set up a FreeSWITCH box with a Sangoma B700 connected to 2 x 
ISDN2e lines (each 2B+D, so 4 voice channels) in UK, so supplied by BT 
Openreach.  There are a number of anomalies that I am trying to solve.  
[One of these is that 1 line is dead, but that is for BT Openreach to 
resolve rather than anyone on this list.]

I have a few questions, but as they are all related, I hope you don't 
mind them in one post.  Some basics first.  The box is an Atom dual-core 
with 2GB of memory and a Sangoma B700 card.  It's built with Ubuntu 
10.04.2 LTS server 64-bit, patched and up to date, and also runs dhcpd, 
lighttpd and sshd.  I built the Sangoma ISDN libraries and FreeSWITCH 
using the latest git versions I could ('make current' about 2 weeks 
ago).  We have incoming and outgoing calls working, but some incoming 
calls ring in the caller's ear, but nothing appears on the FreeSWITCH 
console, and others fail.  Rebooting the server cures this.  We have had 
problems with lines being disconnected and then reconnected, and it 
seems that FreeTDM/wanrouter/whatever doesn't recognise previously 
disconnected lines coming back into use, because a reboot finds lines 
that were previously reported disconnected ('wanrouter status' or 'ftdm 
list').  We have ongoing problems with one line that is 'disconnected' 
(wanpipe2/wp2) but the queries I am describing below apply equally when 
both lines are connected and working.

1. There seems to be a lot of ISDN 'chatter' with channels going up and 
down all the time even when the switch is completely idle.  Is this 
normal?  Here is my /log 7 showing two of the cycles (which appear to be 
about 50s apart):

2011-06-06 22:01:54.992467 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:681 
[SNGISDN Q921] wp1: Protocol: Data Link connection DN(4): Disconnect 
initiated(263)
2011-06-06 22:02:29.952464 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:748 
[SNGISDN Q931] s1: Interface: Down(261): Dchan(285)
2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43 
[s1c1][1:1] Signalling link status changed to DOWN
2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43 
[s1c2][1:2] Signalling link status changed to DOWN
2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43 
[s1c3][1:3] Signalling link status changed to DOWN
2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:200 
[s1c1][1:1] Setting availability rate to:5
2011-06-06 22:02:29.952464 [DEBUG] mod_freetdm.c:2268 got clear channel 
sig [SIGSTATUS_CHANGED]
2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:186 
[s1c1][1:1] Setting availability rate to:5
2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:200 
[s1c2][1:2] Setting availability rate to:5
2011-06-06 22:02:29.952464 [INFO] mod_freetdm.c:2345 1:1 signalling 
changed to :DOWN
2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:186 
[s1c2][1:2] Setting availability rate to:5
2011-06-06 22:02:29.952464 [DEBUG] mod_freetdm.c:2268 got clear channel 
sig [SIGSTATUS_CHANGED]
2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:200 
[s1c3][1:3] Setting availability rate to:5
2011-06-06 22:02:29.952464 [INFO] mod_freetdm.c:2345 1:2 signalling 
changed to :DOWN
2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:186 
[s1c3][1:3] Setting availability rate to:5
2011-06-06 22:02:29.952464 [DEBUG] mod_freetdm.c:2268 got clear channel 
sig [SIGSTATUS_CHANGED]
2011-06-06 22:02:29.952464 [INFO] mod_freetdm.c:2345 1:3 signalling 
changed to :DOWN
2011-06-06 22:02:29.952464 [INFO] ftmod_sangoma_isdn_stack_rcv.c:631 
Received RESTART CFM (dChan:1 ces:0 type:1)
2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:1179 
Receved RESTART, but Restart Indicator IE not present
2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:681 
[SNGISDN Q921] wp1: Protocol: Data Link connection UP(3): UA frame with 
F-bit = 1(258)
2011-06-06 22:02:29.972464 [INFO] ftmod_sangoma_isdn_stack_rcv.c:631 
Received RESTART CFM (dChan:1 ces:0 type:0)
2011-06-06 22:02:29.972464 [INFO] ftmod_sangoma_isdn_stack_rcv.c:729 
[SNGISDN Q931] s1: Interface: UP(260): Dchan(285)
2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43 
[s1c1][1:1] Signalling link status changed to UP
2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43 
[s1c2][1:2] Signalling link status changed to UP
2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43 
[s1c3][1:3] Signalling link status changed to UP
2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:200 
[s1c1][1:1] Setting availability rate to:10
2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:186 
[s1c1][1:1] Setting availability rate to:10
2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:200 
[s1c2][1:2] Setting availability rate to:10
2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:186 
[s1c2][1:2] Setting availability rate to:10
2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:200 
[s1c3][1:3] Setting availability rate to:10
2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:186 
[s1c3][1:3] Setting availability rate to:10
2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:1179 
Receved RESTART, but Restart Indicator IE not present
2011-06-06 22:02:29.972464 [DEBUG] mod_freetdm.c:2268 got clear channel 
sig [SIGSTATUS_CHANGED]
2011-06-06 22:02:29.972464 [INFO] mod_freetdm.c:2345 1:1 signalling 
changed to :UP
2011-06-06 22:02:29.972464 [DEBUG] mod_freetdm.c:2268 got clear channel 
sig [SIGSTATUS_CHANGED]
2011-06-06 22:02:29.972464 [INFO] mod_freetdm.c:2345 1:2 signalling 
changed to :UP
2011-06-06 22:02:29.972464 [DEBUG] mod_freetdm.c:2268 got clear channel 
sig [SIGSTATUS_CHANGED]
2011-06-06 22:02:29.972464 [INFO] mod_freetdm.c:2345 1:3 signalling 
changed to :UP
2011-06-06 22:03:19.912462 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:681 
[SNGISDN Q921] wp1: Protocol: Data Link connection DN(4): Disconnect 
initiated(263)
2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:681 
[SNGISDN Q921] wp1: Protocol: Data Link connection UP(3): UA frame with 
F-bit = 1(258)
2011-06-06 22:03:19.932483 [INFO] ftmod_sangoma_isdn_stack_rcv.c:631 
Received RESTART CFM (dChan:1 ces:0 type:0)
2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:1179 
Receved RESTART, but Restart Indicator IE not present
2011-06-06 22:03:19.932483 [INFO] ftmod_sangoma_isdn_stack_rcv.c:729 
[SNGISDN Q931] s1: Interface: UP(260): Dchan(285)
2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43 
[s1c1][1:1] Signalling link status changed to UP
2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43 
[s1c2][1:2] Signalling link status changed to UP
2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43 
[s1c3][1:3] Signalling link status changed to UP
2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:200 
[s1c1][1:1] Setting availability rate to:10
2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:186 
[s1c1][1:1] Setting availability rate to:10
2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:200 
[s1c2][1:2] Setting availability rate to:10
2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:186 
[s1c2][1:2] Setting availability rate to:10
2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:200 
[s1c3][1:3] Setting availability rate to:10
2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:186 
[s1c3][1:3] Setting availability rate to:10
2011-06-06 22:03:19.932483 [DEBUG] mod_freetdm.c:2268 got clear channel 
sig [SIGSTATUS_CHANGED]
2011-06-06 22:03:19.932483 [INFO] mod_freetdm.c:2345 1:1 signalling 
changed to :UP
2011-06-06 22:03:19.932483 [DEBUG] mod_freetdm.c:2268 got clear channel 
sig [SIGSTATUS_CHANGED]
2011-06-06 22:03:19.932483 [INFO] mod_freetdm.c:2345 1:2 signalling 
changed to :UP
2011-06-06 22:03:19.932483 [DEBUG] mod_freetdm.c:2268 got clear channel 
sig [SIGSTATUS_CHANGED]
2011-06-06 22:03:19.932483 [INFO] mod_freetdm.c:2345 1:3 signalling 
changed to :UP
freeswitch at internal> ftdm list
+OK
span: 1 (wp1)
type: Sangoma (ISDN)
physical_status: ok
signaling_status: UP
chan_count: 3
dialplan: XML
context: public
dial_regex:
fail_dial_regex:
hold_music:
analog_options: none
+OK
span: 2 (wp2)
type: Sangoma (ISDN)
physical_status: alarmed
signaling_status: DOWN
chan_count: 3
dialplan: XML
context: public
dial_regex:
fail_dial_regex:
hold_music:
analog_options: none
+OK
span: 3 (FXS)
type: analog
physical_status: ok
signaling_status: UP
chan_count: 2
dialplan: XML
context: default
dial_regex:
fail_dial_regex:
hold_music:
analog_options: none


# wanrouter status

Devices currently active:
     wanpipe1 wanpipe2 wanpipe3


Wanpipe Config:

Device name | Protocol Map | Adapter  | IRQ | Slot/IO | If's | CLK | 
Baud rate |
wanpipe1    | N/A          | A500/B700| 20  | 0       | 1    | N/A | 
0         |
wanpipe2    | N/A          | A500/B700| 20  | 0       | 1    | N/A | 
0         |
wanpipe3    | N/A          | A200/A400/B600/B700/B800| 20  | 0       | 
1    | N/A | 0         |

Wanrouter Status:

Device name | Protocol | Station | Status        |
wanpipe1    | AFT ISDN | N/A     | Connected     |
wanpipe2    | AFT ISDN | N/A     | Disconnected  |
wanpipe3    | A-ANALOG | N/A     | Connected     |


# cat /etc/wanpipe/wanrouter.rc
#!/bin/sh
# .. comments snipped ...
ROUTER_BOOT=YES
WAN_CONF_DIR=/etc/wanpipe
WAN_INTR_DIR=/etc/wanpipe/interfaces
WAN_BIN_DIR=/usr/sbin
WAN_LOG=/var/log/wanrouter
WAN_LOCK=/var/lock/wanrouter
WAN_LOCK_DIR=/var/lock
WAN_IP_FORWARD=NO
NEW_IF_TYPE=NO
WAN_LIB_DIR=/etc/wanpipe/lib
WAN_ADSL_LIST=/etc/wanpipe/wan_adsl.list
WAN_ANNEXG_LOAD=NO
WAN_SCTP_LOAD=NO
WAN_LIP_LOAD=NO
WAN_DYN_WANCONFIG=NO
WAN_SCRIPTS_DIR=/etc/wanpipe/scripts
WAN_FIRMWARE_DIR=/etc/wanpipe/firmware
WAN_DEVICES_REV_STOP_ORDER=YES
WAN_DEVICES="wanpipe1 wanpipe2 wanpipe3 "

# cat /etc/wanpipe/wanpipe1.conf
#================================================
# WANPIPE1 Configuration File
#================================================
#
# Note: This file was generated automatically
#       by /usr/local/sbin/setup-sangoma program.
#
#       If you want to edit this file, it is
#       recommended that you use wancfg program
#       to do so.
#================================================
# Sangoma Technologies Inc.
#================================================




[devices]
wanpipe1 = WAN_AFT_ISDN_BRI, Comment

[interfaces]
w1g1 = wanpipe1, , TDM_VOICE_API, Comment

[wanpipe1]
CARD_TYPE     = AFT
S514CPU     = A
CommPort     = PRI
AUTO_PCISLOT     = NO
PCISLOT     = 0
PCIBUS      = 5
FE_MEDIA    = BRI
FE_LINE        = 1
TDMV_LAW    = ALAW
RM_BRI_CLOCK_MASTER    = NO
MTU         = 1500
UDPPORT     = 9000
TTL        = 255
IGNORE_FRONT_END = NO
TDMV_SPAN    = 1
TE_AIS_MAINTENANCE = NO            #NO: defualt  YES: Start port in AIS 
Blue Alarm and keep line down
                                 #wanpipemon -i w1g1 -c Ttx_ais_off to 
disable AIS maintenance mode
                                 #wanpipemon -i w1g1 -c Ttx_ais_on to 
enable AIS maintenance mode
TDMV_HW_DTMF        = YES        # YES: receive dtmf events from hardware
TDMV_HW_FAX_DETECT      = YES     # YES: receive fax 1100hz events from 
hardware
HWEC_OPERATION_MODE     = OCT_NORMAL    # OCT_NORMAL: echo cancelation 
enabled with nlp (default)
                                         # OCT_SPEECH: improves software 
tone detection by disabling NLP (echo possible)
                                         # OCT_NO_ECHO:disables echo 
cancelation but allows VQE/tone functions.
HWEC_DTMF_REMOVAL       = NO    # NO: default  YES: remove dtmf out of 
incoming media (must have hwdtmf enabled)
HWEC_NOISE_REDUCTION    = NO    # NO: default  YES: reduces noise on the 
line - could break fax
HWEC_ACUSTIC_ECHO       = NO    # NO: default  YES: enables acustic echo 
cancelation
HWEC_NLP_DISABLE        = NO    # NO: default  YES: guarantees software 
tone detection (possible echo)
HWEC_TX_AUTO_GAIN       = 0     # 0: disable   -40-0: default tx audio 
level to be maintained (-20 default)
HWEC_RX_AUTO_GAIN       = 0     # 0: disable   -40-0: default tx audio 
level to be maintained (-20 default)
HWEC_TX_GAIN            = 0     # 0: disable   -24-24: db values to be 
applied to tx signal
HWEC_RX_GAIN            = 0     # 0: disable   -24-24: db values to be 
applied to tx signal

[w1g1]
ACTIVE_CH    = ALL
TDMV_HWEC    = YES
MTU         = 80

  ... and wanpipe2 was also automatically generated so just has '2' for 
FE_LINE and TDMV_SPAN instead of '1'.

# cat /usr/local/freeswitch/conf/freetdm.conf
[span wanpipe wp1]
trunk_type => bri
group=1
b-channel => 1:1-2
d-channel => 1:3

[span wanpipe wp2]
trunk_type => bri
group=1
b-channel => 2:1-2
d-channel => 2:3

[span wanpipe FXS]
name => freetdm
trunk_type => fxs
group => grp2
fxs-channel => 3:1

trunk_type => fxs
group => grp2
fxs-channel => 3:2

# cat /usr/local/freeswitch/conf/autoload_configs/freetdm.conf.xml
<configuration name="freetdm.conf" description="Freetdm Configuration">
<settings>
<param name="debug" value="0"/>
<!--<param name="hold-music" value="$${moh_uri}"/>-->
<!--<param name="enable-analog-option" value="call-swap"/>-->
<!--<param name="enable-analog-option" value="3-way"/>-->
</settings>

<config_profiles>

<profile name="my_bri_te_1">
<param name="switchtype" value="euroisdn" />
<param name="interface" value="cpe"/>
<param name="facility" value="no" />
</profile>

</config_profiles>

<sangoma_bri_spans>

<span name="wp1" cfgprofile="my_bri_te_1">
<param name="dialplan" value="XML"/>
<param name="context" value="public"/>
</span>

<span name="wp2" cfgprofile="my_bri_te_1">
<param name="dialplan" value="XML"/>
<param name="context" value="public"/>
</span>

</sangoma_bri_spans>


<analog_spans>

<span name="FXS">
<!--<param name="hold-music" value="$${moh_uri}"/>-->
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
<!-- regex to stop dialing when it matches -->
<!--<param name="dial-regex" value="5555"/>-->
<!-- regex to stop dialing when it does not match -->
<!--<param name="fail-dial-regex" value="^5"/>-->
</span>

</analog_spans>

</configuration>

What other config files do I need to supply? I don't want to deluge the 
list any more in my first post!

2. We have 2 Single Number DDI numbers configured over the 4 channels.  
I want to set up outgoing calls so that they can appear to come from 
either of the two numbers, but at present the outgoing CLI appears to be 
overridden by the telco (BT) to only use one of the numbers.  Has anyone 
got this working in UK, and what is the format for the 
outbound_caller_id_number: last 6 digits or full 11 digits?  I note that 
the inbound called number is only the last 6 digits.

3. I have built FreeSWITCH from git and installed at /usr/local/... and 
then followed the steps on the Ubuntu page in the Wiki to set up the run 
control scripts, etc, and run FS non-root as freeswitch:daemon.  With 
FreeTDM, I have discovered that the /dev/wan* devices are owned by 
root:root, and so are inaccessible to FS running as non-root.  So for 
now I have added a line in /etc/init.d/freeswitch to 'chgrp freeswitch 
/dev/wan*'.  This is not the most elegant solution, because 'wanrouter 
restart' (which seems to be my best friend at the moment) resets the 
ownership to root:root.  I have tried grepping to see where the mknods 
are for these devices, but have been unsuccessful.  Is there a better 
place to 'permanently' change the device ownership?

Thanks for all the great support I have already got just from editing my 
Wiki User page; this is a friendly group!

John


From fredyg1965 at gmail.com  Tue Jun  7 02:01:35 2011
From: fredyg1965 at gmail.com (Fredy Gonzales)
Date: Mon, 6 Jun 2011 17:01:35 -0500
Subject: [Freeswitch-users] Speech Recognition: Australian English
References: <4DEA47E7.8010306@vestec.com>
	<5391553A31334E0587B99203F8E78C96@gmail.com>
Message-ID: <87244E7710894AF1B59375BBE784A890@gcg.com.pe>

Plans to add Spanish language?
  ----- Original Message ----- 
  From: Seven Du 
  To: FreeSWITCH Users Help 
  Sent: Saturday, June 04, 2011 6:24 PM
  Subject: Re: [Freeswitch-users] Speech Recognition: Australian English


  Interested in dose it offer trial version?

  -- 
  Seven Du

  About: http://about.me/dujinfang
  Blog: http://www.dujinfang.com
  Proj:  http://www.freeswitch.org.cn
  Sent with Sparrow

  On Saturday, June 4, 2011 at 10:57 PM, Kashif Kahn wrote:

    Hello Everyone,

    Vestec ASR engine now supports speech recognition in Australian English. 
    (We also support American English, Indian English, and Chinese 
    Mandarin). A starter kit is available for $25: 
    http://www.vestec.com/products

    Please note that Vestec offers the best deal around for enabling 
    sophisticated speech recognition with command-and-control type IVR 
    applications by offering a high accuracy, standards based speech engine 
    at a fraction of the cost of conventional ASR vendors. Feel free to 
    contact me with any questions or concerns.

    Regards,
    -Kashif


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





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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From krice at freeswitch.org  Tue Jun  7 09:27:07 2011
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 07 Jun 2011 00:27:07 -0500
Subject: [Freeswitch-users] Freeswitch 1.0.6 doesn't have mod_hash.c?
In-Reply-To: <4DED9F50.50109@tagnet.ru>
Message-ID: <CA1320DB.170D0%krice@freeswitch.org>

How about quit using 1.0.6 and update to current...


On 6/6/11 10:47 PM, "Boris Kovalenko" <boris at tagnet.ru> wrote:

>    Hello!
>  
>      It is in source tree now. Please read Changelog
>  
>  
>> but my fs running time complains : switch_core_session.c:1731 Invalid
>> Application hash. I couldn't find mod_hash.c in src package...
>> 
>>  
>>  
>> any idea?
>>  
>> 
>>  
>>  
>> thanks!
>>  
>> 
>>  
>>  
>>  
>>  
>> 
>>  
>>  
>>  
>>  
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>  
>  
>  
>  

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From steveayre at gmail.com  Tue Jun  7 10:35:56 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 7 Jun 2011 07:35:56 +0100
Subject: [Freeswitch-users] Freeswitch 1.0.6 doesn't have mod_hash.c?
In-Reply-To: <BANLkTik5eLVYmcQfcDBjD6natOdtsVVLDFD5jSxyCWAybG6-SQ@mail.gmail.com>
References: <BANLkTik5eLVYmcQfcDBjD6natOdtsVVLDFD5jSxyCWAybG6-SQ@mail.gmail.com>
Message-ID: <BANLkTimzz+O4KC1m6690X6QnTmdOBLuoEg@mail.gmail.com>

It's supplied by mod_limit. That module has been removed from the latest
versions due to a large reimplementation of the Limit system (it's now in
the core with a new API so that you can have pluggable modules providing
custom backends for Limit).

You could load mod_limit for now, but you'll have to switch to loading
mod_hash when you upgrade. You're best off developing with the latest
version to save you some extra work when you do eventually upgrade.

Plenty of bugs have been fixed in git head since 1.0.6 too.

-Steve


On 7 June 2011 00:10, jesse <chat2jesse at gmail.com> wrote:

> but my fs running time complains : switch_core_session.c:1731 Invalid
> Application hash. I couldn't find mod_hash.c in src package...
>
> any idea?
>
> thanks!
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sunwood360 at gmail.com  Tue Jun  7 10:37:15 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Mon, 6 Jun 2011 23:37:15 -0700
Subject: [Freeswitch-users] Freeswitch 1.0.6 doesn't have mod_hash.c?
In-Reply-To: <CA1320DB.170D0%krice@freeswitch.org>
References: <4DED9F50.50109@tagnet.ru>
	<CA1320DB.170D0%krice@freeswitch.org>
Message-ID: <BANLkTinfm+miJ2a=vNcMfVV1Hc3TAOcaEA@mail.gmail.com>

Sure, I am just curious about this error message since there is no
mod_hash.c in 1.0.6.
On Jun 6, 2011 10:29 PM, "Ken Rice" <krice at freeswitch.org> wrote:
> How about quit using 1.0.6 and update to current...
>
>
> On 6/6/11 10:47 PM, "Boris Kovalenko" <boris at tagnet.ru> wrote:
>
>> Hello!
>>
>> It is in source tree now. Please read Changelog
>>
>>
>>> but my fs running time complains : switch_core_session.c:1731 Invalid
>>> Application hash. I couldn't find mod_hash.c in src package...
>>>
>>>
>>>
>>> any idea?
>>>
>>>
>>>
>>>
>>> thanks!
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>
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From p.devarahalli at alcatel-lucent.com  Tue Jun  7 09:16:51 2011
From: p.devarahalli at alcatel-lucent.com (DEVARAHALLI, PRAKASH (PRAKASH))
Date: Tue, 7 Jun 2011 10:46:51 +0530
Subject: [Freeswitch-users] FS does not uses the IP/PORT resolved using STUN.
Message-ID: <7E79416ABDDDFC4E8C85571D0B5CAB3F138AE81452@INBANSXCHMBSA1.in.alcatel-lucent.com>

I configured FS to use the STUN server to find the public IP address used by the NAT.
FS sends the request to STUN and receives the reply from the STUN Server , But it does
not uses this info ( IP and port obtained from STUN)  neither while negotiation ( in SDP)
nor in the RTP.


My testbed:
-----------

< FreeSwitch > ---- < NAT > ---+----<NAT>--- < SIP Phone Client >
                                             |
                                             |
                               [ STUN Server]


File conf/sip_profiles/internal.xml has below configuration.

   <param name="ext-rtp-ip" value="stun:120.120.120.120"/> ; This is the local STUN server.
   <param name="ext-sip-ip" value="stun:120.120.120.120"/>
   <param name="stun-auto-disable" value="false"/>
   <param name="rtp-ip" value="auto"/>

Thanks
Prakash


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From steveayre at gmail.com  Tue Jun  7 11:51:47 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 7 Jun 2011 08:51:47 +0100
Subject: [Freeswitch-users] Freeswitch 1.0.6 doesn't have mod_hash.c?
In-Reply-To: <BANLkTinfm+miJ2a=vNcMfVV1Hc3TAOcaEA@mail.gmail.com>
References: <4DED9F50.50109@tagnet.ru> <CA1320DB.170D0%krice@freeswitch.org>
	<BANLkTinfm+miJ2a=vNcMfVV1Hc3TAOcaEA@mail.gmail.com>
Message-ID: <570EC7DE-9D4A-48A4-8B8B-9F13FD51C033@gmail.com>

Well more specifically, any module can register an application with any name it likes. In this case you're trying to call an application named hash which hasn't been loaded.

In 1.0.6 that is registered by mod_limit, in more recent versions it's provided by the newer mod_hash which isn't in the older versions.

Steve on iPhone

On 7 Jun 2011, at 07:37, envelopes envelopes <sunwood360 at gmail.com> wrote:

> Sure, I am just curious about this error message since there is no mod_hash.c in 1.0.6.
> 
> On Jun 6, 2011 10:29 PM, "Ken Rice" <krice at freeswitch.org> wrote:
> > How about quit using 1.0.6 and update to current...
> > 
> > 
> > On 6/6/11 10:47 PM, "Boris Kovalenko" <boris at tagnet.ru> wrote:
> > 
> >> Hello!
> >> 
> >> It is in source tree now. Please read Changelog
> >> 
> >> 
> >>> but my fs running time complains : switch_core_session.c:1731 Invalid
> >>> Application hash. I couldn't find mod_hash.c in src package...
> >>> 
> >>> 
> >>> 
> >>> any idea?
> >>> 
> >>> 
> >>> 
> >>> 
> >>> thanks!
> >>> 
> >>> 
> >>> 
> >>> 
> >>> 
> >>> 
> >>> 
> >>> 
> >>> 
> >>> 
> >>> 
> >>> 
> >>> 
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>> 
> >> 
> >> 
> >> 
> > 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From p.devarahalli at alcatel-lucent.com  Tue Jun  7 11:53:32 2011
From: p.devarahalli at alcatel-lucent.com (DEVARAHALLI, PRAKASH (PRAKASH))
Date: Tue, 7 Jun 2011 13:23:32 +0530
Subject: [Freeswitch-users] FS does not uses the IP/PORT resolved using STUN.
Message-ID: <7E79416ABDDDFC4E8C85571D0B5CAB3F138AE814F1@INBANSXCHMBSA1.in.alcatel-lucent.com>

I configured FS to use the STUN server to find the public IP address used by the NAT.
FS sends the request to STUN and receives the reply from the STUN Server , But it does
not uses this info ( IP and port obtained from STUN)  neither while negotiation ( in SDP)
nor in the RTP.


My testbed:
-----------

< FreeSwitch > ---- < NAT > ---+----<NAT>--- < SIP Phone Client >
                                             |
                                             |
                               [ STUN Server]


File conf/sip_profiles/internal.xml has below configuration.

   <param name="ext-rtp-ip" value="stun:120.120.120.120"/> ; This is the local STUN server.
   <param name="ext-sip-ip" value="stun:120.120.120.120"/>
   <param name="stun-auto-disable" value="false"/>
   <param name="rtp-ip" value="auto"/>

Thanks
Prakash


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From grsingh750 at gmail.com  Tue Jun  7 13:14:24 2011
From: grsingh750 at gmail.com (guru singh)
Date: Tue, 7 Jun 2011 14:44:24 +0530
Subject: [Freeswitch-users] FS does not uses the IP/PORT resolved using
	STUN.
In-Reply-To: <7E79416ABDDDFC4E8C85571D0B5CAB3F138AE814F1@INBANSXCHMBSA1.in.alcatel-lucent.com>
References: <7E79416ABDDDFC4E8C85571D0B5CAB3F138AE814F1@INBANSXCHMBSA1.in.alcatel-lucent.com>
Message-ID: <BANLkTikmdM4qQ7x9Z753KenzxwmQwTFbDw@mail.gmail.com>

Hi,

I've never used STUN but if you have a router that supports UPnP/PMP,
you could try configuring your sofia profile to use auto-nat , it
usually does the job.

Regards,
guru

On Tue, Jun 7, 2011 at 1:23 PM, DEVARAHALLI, PRAKASH (PRAKASH)
<p.devarahalli at alcatel-lucent.com> wrote:
> I configured FS to use the STUN server to find the public IP address used by
> the NAT.
> FS sends the request to STUN and receives the reply from the STUN Server ,
> But it does
> not uses this info ( IP and port obtained from STUN)? neither while
> negotiation ( in SDP)
> nor in the RTP.
>
>
> My testbed:
> -----------
>
> < FreeSwitch > ---- < NAT > ---+----<NAT>--- < SIP Phone Client >
> ???????????????????????????????????????????? |
> ???????????????????????????????????????????? |
> ??????????????????????????????? [ STUN Server]
>
>
> File conf/sip_profiles/internal.xml has below configuration.
>
> ?? <param name="ext-rtp-ip" value="stun:120.120.120.120"/> ; This is the
> local STUN server.
> ?? <param name="ext-sip-ip" value="stun:120.120.120.120"/>
> ?? <param name="stun-auto-disable" value="false"/>
> ?? <param name="rtp-ip" value="auto"/>
>
> Thanks
> Prakash
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From david.ponzone at ipeva.fr  Tue Jun  7 16:11:00 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 7 Jun 2011 14:11:00 +0200
Subject: [Freeswitch-users] Questions
In-Reply-To: <SNT110-W43983F1D6DACAB71BE8CE1CD600@phx.gbl>
References: <SNT110-W43983F1D6DACAB71BE8CE1CD600@phx.gbl>
Message-ID: <176BC431-D84F-4E75-8CAA-7015AB3AFCB2@ipeva.fr>

Fabian,

I would really recommend staying away from H323 on FS.
As far as I know, both modules are far from perfect, and you would have to be closely in touch with the devs to get them working as you need.
I know people who went for Yate, just to convert H323 into SIP, and then FS.

About the virtual interfaces, FS does not have to deal with that.
It's the network layer of the Linux kernel which decides which interface to use.

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

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

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




Le 06/06/2011 ? 21:16, Fabian Borot a ?crit :

> Hello
> 
> We are giving Freeswitch a try in our organization to provide H323 <-> SIP functionality. We have 2 main questions/requests before starting installing/configuring:
> 
> 1- What would be the most recommended module for H323? I see there are two [opal and mod_h323] both both say that are in development stages. 
> 
> 2- Does FS fully supports virtual interfaces [eth0:1]? I mean: 
>         2- 1 listening on several virtual interfaces,
>         2-2 replying from the virtual interface that the request came in instead of replying from the primary interface
>         2-3 and also we need to specify the interface to send the call out instead of using the primary interface
> 
> Thank you
> 
> Fabian
> 
> 
> 
> 
>  
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From fborot at hotmail.com  Tue Jun  7 16:37:39 2011
From: fborot at hotmail.com (Fabian Borot)
Date: Tue, 7 Jun 2011 08:37:39 -0400
Subject: [Freeswitch-users] Questions
In-Reply-To: <SNT110-W43983F1D6DACAB71BE8CE1CD600@phx.gbl>
References: <SNT110-W43983F1D6DACAB71BE8CE1CD600@phx.gbl>
Message-ID: <SNT110-W4675FBABE69CB3C7374482CD630@phx.gbl>



Thank you David for your reply. I guess I'll do that.

 



From: fborot at hotmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Questions
Date: Mon, 6 Jun 2011 15:16:25 -0400








Hello

We are giving Freeswitch a try in our organization to provide H323 <-> SIP functionality. We have 2 main questions/requests before starting installing/configuring:

1- What would be the most recommended module for H323? I see there are two [opal and mod_h323] both both say that are in development stages. 

2- Does FS fully supports virtual interfaces [eth0:1]? I mean: 
        2- 1 listening on several virtual interfaces,
        2-2 replying from the virtual interface that the request came in instead of replying from the primary interface
        2-3 and also we need to specify the interface to send the call out instead of using the primary interface

Thank you

Fabian





 

 		 	   		  
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From benkokakao at gmail.com  Tue Jun  7 16:39:10 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Tue, 7 Jun 2011 14:39:10 +0200
Subject: [Freeswitch-users] Questions
In-Reply-To: <SNT110-W43983F1D6DACAB71BE8CE1CD600@phx.gbl>
References: <SNT110-W43983F1D6DACAB71BE8CE1CD600@phx.gbl>
Message-ID: <BANLkTi=Pco4eH7jCmhBerM3UYRAnE-7NBg@mail.gmail.com>

> 2- Does FS fully supports virtual interfaces [eth0:1]? I mean:
> ??????? 2- 1 listening on several virtual interfaces,
> ? ? ??? 2-2 replying from the virtual interface that the request came in
> instead of replying from the primary interface
> ? ? ? ? 2-3 and also we need to specify the interface to send the call out
> instead of using the primary interface


You don't define interfaces but the ip-adresses where freeswitch
listens on(take a look at external_sip_ip, external_rtp_ip, domain and
local_ip_v4 and sip_profiles).
So yes, virtual interfaces can be served, several ip's are possible
too, but you have to define each of them in separate sip-profiles.


From steveayre at gmail.com  Tue Jun  7 18:02:50 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 7 Jun 2011 15:02:50 +0100
Subject: [Freeswitch-users] Questions
In-Reply-To: <176BC431-D84F-4E75-8CAA-7015AB3AFCB2@ipeva.fr>
References: <SNT110-W43983F1D6DACAB71BE8CE1CD600@phx.gbl>
	<176BC431-D84F-4E75-8CAA-7015AB3AFCB2@ipeva.fr>
Message-ID: <BANLkTim7M-57LWJL8LK0npb3ag2n+2x2fA@mail.gmail.com>

I couldn't get mod_opal to compile (probably because I was on debian lenny
at the time). mod_h323 worked better, but did have stability issues. We've
actually moved to being 100% SIP now though and it's some time since I tried
using either.

We used to use Yate as a H323->SIP proxy, the configuration details are
here:
http://yate.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy

-Steve


On 7 June 2011 13:11, David Ponzone <david.ponzone at ipeva.fr> wrote:

> Fabian,
>
> I would really recommend staying away from H323 on FS.
> As far as I know, both modules are far from perfect, and you would have to
> be closely in touch with the devs to get them working as you need.
> I know people who went for Yate, just to convert H323 into SIP, and then
> FS.
>
> About the virtual interfaces, FS does not have to deal with that.
> It's the network layer of the Linux kernel which decides which interface to
> use.
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 06/06/2011 ? 21:16, Fabian Borot a ?crit :
>
> Hello
>
> We are giving Freeswitch a try in our organization to provide H323 <-> SIP
> functionality. We have 2 main questions/requests before starting
> installing/configuring:
>
> 1- What would be the most recommended module for H323? I see there are two
> [opal and mod_h323] both both say that are in development stages.
>
> 2- Does FS fully supports virtual interfaces [eth0:1]? I mean:
>         2- 1 listening on several virtual interfaces,
>         2-2 replying from the virtual interface that the request came in
> instead of replying from the primary interface
>         2-3 and also we need to specify the interface to send the call out
> instead of using the primary interface
>
> Thank you
>
> Fabian
>
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Tue Jun  7 18:04:11 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 7 Jun 2011 15:04:11 +0100
Subject: [Freeswitch-users] Questions
In-Reply-To: <SNT110-W43983F1D6DACAB71BE8CE1CD600@phx.gbl>
References: <SNT110-W43983F1D6DACAB71BE8CE1CD600@phx.gbl>
Message-ID: <BANLkTin=8nRF1bL582B=p=B47VqdCyOOzQ@mail.gmail.com>

>
> 2- Does FS fully supports virtual interfaces [eth0:1]? I mean:
>         2- 1 listening on several virtual interfaces,
>         2-2 replying from the virtual interface that the request came in
> instead of replying from the primary interface
>         2-3 and also we need to specify the interface to send the call out
> instead of using the primary interface


FS will bind to the IP rather than the device. That does mean it'll always
send/receive using the IP you set - is that what you're asking?

-Steve


On 6 June 2011 20:16, Fabian Borot <fborot at hotmail.com> wrote:

>  Hello
>
> We are giving Freeswitch a try in our organization to provide H323 <-> SIP
> functionality. We have 2 main questions/requests before starting
> installing/configuring:
>
> 1- What would be the most recommended module for H323? I see there are two
> [opal and mod_h323] both both say that are in development stages.
>
> 2- Does FS fully supports virtual interfaces [eth0:1]? I mean:
>         2- 1 listening on several virtual interfaces,
>         2-2 replying from the virtual interface that the request came in
> instead of replying from the primary interface
>         2-3 and also we need to specify the interface to send the call out
> instead of using the primary interface
>
> Thank you
>
> Fabian
>
>
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rajesh.npnr at yahoo.com  Tue Jun  7 18:09:57 2011
From: rajesh.npnr at yahoo.com (rex.alex)
Date: Tue, 7 Jun 2011 07:09:57 -0700 (PDT)
Subject: [Freeswitch-users] mod_com_g729 version for freeswitch 1.0.6
Message-ID: <1307455797128-6449771.post@n2.nabble.com>

Hello,

I just purchased a g729 license for testing purpose and based on
http://files.freeswitch.org/g729/INSTALL.txt file, installed
fsg729-194-installer. My freeswitch version is 1.0.6. I have unloaded the
mod_g729 and when I am trying to load mod_com_g729, it shows the following
error. Please assist.

freeswitch at internal> load mod_com_g729
-ERR [module load file routine returned an error]
freeswitch at internal> 2011-06-07 03:03:08.294418 [CRIT]
switch_loadable_module.c:882 Error Loading module
/usr/local/freeswitch/mod/mod_com_g729.so
**Trying to load an out of date module, please rebuild the module.**

Thanks,
Rex.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-com-g729-version-for-freeswitch-1-0-6-tp6449771p6449771.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Tue Jun  7 18:46:48 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 7 Jun 2011 15:46:48 +0100
Subject: [Freeswitch-users] mod_com_g729 version for freeswitch 1.0.6
In-Reply-To: <1307455797128-6449771.post@n2.nabble.com>
References: <1307455797128-6449771.post@n2.nabble.com>
Message-ID: <42319CAF-2786-40A4-895F-BEC9C273A22E@gmail.com>

It won't work with 1.0.6, you'll need a newer version of FS. Please upgrade to git head.

Steve on iPhone

On 7 Jun 2011, at 15:09, "rex.alex" <rajesh.npnr at yahoo.com> wrote:

> Hello,
> 
> I just purchased a g729 license for testing purpose and based on
> http://files.freeswitch.org/g729/INSTALL.txt file, installed
> fsg729-194-installer. My freeswitch version is 1.0.6. I have unloaded the
> mod_g729 and when I am trying to load mod_com_g729, it shows the following
> error. Please assist.
> 
> freeswitch at internal> load mod_com_g729
> -ERR [module load file routine returned an error]
> freeswitch at internal> 2011-06-07 03:03:08.294418 [CRIT]
> switch_loadable_module.c:882 Error Loading module
> /usr/local/freeswitch/mod/mod_com_g729.so
> **Trying to load an out of date module, please rebuild the module.**
> 
> Thanks,
> Rex.
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-com-g729-version-for-freeswitch-1-0-6-tp6449771p6449771.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From yungwei at resolvity.com  Tue Jun  7 19:05:50 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Tue, 7 Jun 2011 11:05:50 -0400
Subject: [Freeswitch-users] A DTMF issue
In-Reply-To: <BANLkTi=vcvoLy=eU7f_=0bGCJ=2K2NjGfw@mail.gmail.com>
References: <33095823FD21DF429B481B5163264B7950AC585DEE@VMBX102.ihostexchange.net>
	<BANLkTi=vcvoLy=eU7f_=0bGCJ=2K2NjGfw@mail.gmail.com>
Message-ID: <33095823FD21DF429B481B5163264B7950AC5861DC@VMBX102.ihostexchange.net>

That's how it's supposed to work and no workaround?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: Friday, June 03, 2011 5:46 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] A DTMF issue

You should be checking session.ready() in that loop, as it is I think you'll find the scripts are never ending when the channel is hung up.

streamFile/flushDigits probably return instantly, but then get called again on the next loop iteration. The problem would get slightly worse after each test call since there would be an extra copy of the script running.

Try replacing:
    while(true) {
with:
    while(session.ready()) {

-Steve

On 3 June 2011 23:35, Yungwei Chen wrote:
Hi,

When testing the following javascript program, I notice a problem that it takes longer and longer for freeswitch to respond to DTMF input for some reason.
Please refer to http://pastebin.freeswitch.org/16435 for debug logs.
Notice that the difference in time between any adjacent pairs of the following is increasing as time goes by. Am I missing something here?
       switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
       switch_rtp.c:3302 RTP RECV DTMF 2:960
Thanks.

Here's the javascript program:
var dtmf_digits = "";


function on_dtmf_28(session, type, digits, arg)
{
   if (type == "dtmf")
   {

       dtmf_digits = digits.digit;

       console_log("dtmf_digits=" + dtmf_digits + "\n");

       return false; // barge-in / done
   }
   return true;
}

while (true)
{
       dtmf_digits = "";
       session.flushDigits();
       session.streamFile("/usr/local/freeswitch/sounds/long-prompt.wav", on_dtmf_28, false);

       if(dtmf_digits.length == 0)
       {
           /* no input */
           console_log("no input\n");
           session.speak('flite', 'kal', 'no input!', null);
       }
       else if(dtmf_digits == "1" || dtmf_digits == "2")
       {
               console_log("match "+dtmf_digits+"\n");
               session.speak('flite', 'kal', 'you pressed ' + dtmf_digits+ '!', null);
       }
       else
       {
           /* no match */
           console_log("no match\n");
           session.speak('flite', 'kal', "no match!", null);
       }

}

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From steveayre at gmail.com  Tue Jun  7 19:54:26 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 7 Jun 2011 16:54:26 +0100
Subject: [Freeswitch-users] A DTMF issue
In-Reply-To: <33095823FD21DF429B481B5163264B7950AC5861DC@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950AC585DEE@VMBX102.ihostexchange.net>
	<BANLkTi=vcvoLy=eU7f_=0bGCJ=2K2NjGfw@mail.gmail.com>
	<33095823FD21DF429B481B5163264B7950AC5861DC@VMBX102.ihostexchange.net>
Message-ID: <BANLkTikv5ULzkmnz86TT3zzANR1jb-Us5A@mail.gmail.com>

It's a simple fix. Always check session.ready when you loop. Yes, that's how
it's designed to work.

-Steve



On 7 June 2011 16:05, Yungwei Chen <yungwei at resolvity.com> wrote:

> That's how it's supposed to work and no workaround?
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* Friday, June 03, 2011 5:46 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] A DTMF issue
>
>
>
> You should be checking session.ready() in that loop, as it is I think
> you'll find the scripts are never ending when the channel is hung up.
>
> streamFile/flushDigits probably return instantly, but then get called again
> on the next loop iteration. The problem would get slightly worse after each
> test call since there would be an extra copy of the script running.
>
> Try replacing:
>     while(true) {
> with:
>     while(session.ready()) {
>
> -Steve
>
> On 3 June 2011 23:35, Yungwei Chen wrote:
>
> Hi,
>
> When testing the following javascript program, I notice a problem that it
> takes longer and longer for freeswitch to respond to DTMF input for some
> reason.
> Please refer to http://pastebin.freeswitch.org/16435 for debug logs.
> Notice that the difference in time between any adjacent pairs of the
> following is increasing as time goes by. Am I missing something here?
>        switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels
> 20ms
>        switch_rtp.c:3302 RTP RECV DTMF 2:960
> Thanks.
>
> Here's the javascript program:
> var dtmf_digits = "";
>
>
> function on_dtmf_28(session, type, digits, arg)
> {
>    if (type == "dtmf")
>    {
>
>        dtmf_digits = digits.digit;
>
>        console_log("dtmf_digits=" + dtmf_digits + "\n");
>
>        return false; // barge-in / done
>    }
>    return true;
> }
>
> while (true)
> {
>        dtmf_digits = "";
>        session.flushDigits();
>        session.streamFile("/usr/local/freeswitch/sounds/long-prompt.wav",
> on_dtmf_28, false);
>
>        if(dtmf_digits.length == 0)
>        {
>            /* no input */
>            console_log("no input\n");
>            session.speak('flite', 'kal', 'no input!', null);
>        }
>        else if(dtmf_digits == "1" || dtmf_digits == "2")
>        {
>                console_log("match "+dtmf_digits+"\n");
>                session.speak('flite', 'kal', 'you pressed ' + dtmf_digits+
> '!', null);
>        }
>        else
>        {
>            /* no match */
>            console_log("no match\n");
>            session.speak('flite', 'kal', "no match!", null);
>        }
>
> }
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Jun  7 19:55:20 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 7 Jun 2011 08:55:20 -0700
Subject: [Freeswitch-users] A DTMF issue
In-Reply-To: <33095823FD21DF429B481B5163264B7950AC5861DC@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950AC585DEE@VMBX102.ihostexchange.net>
	<BANLkTi=vcvoLy=eU7f_=0bGCJ=2K2NjGfw@mail.gmail.com>
	<33095823FD21DF429B481B5163264B7950AC5861DC@VMBX102.ihostexchange.net>
Message-ID: <BANLkTikpNteMa6oLw8ZCj=Z9q1XpcGYDkg@mail.gmail.com>

Yes, this is by design.

I think you might be trying to use the wrong tool for the task at hand. What
is the problem that you are trying to solve? What function/feature are you
trying to implement?

-MC

On Tue, Jun 7, 2011 at 8:05 AM, Yungwei Chen <yungwei at resolvity.com> wrote:

> That's how it's supposed to work and no workaround?
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* Friday, June 03, 2011 5:46 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] A DTMF issue
>
>
>
> You should be checking session.ready() in that loop, as it is I think
> you'll find the scripts are never ending when the channel is hung up.
>
> streamFile/flushDigits probably return instantly, but then get called again
> on the next loop iteration. The problem would get slightly worse after each
> test call since there would be an extra copy of the script running.
>
> Try replacing:
>     while(true) {
> with:
>     while(session.ready()) {
>
> -Steve
>
> On 3 June 2011 23:35, Yungwei Chen wrote:
>
> Hi,
>
> When testing the following javascript program, I notice a problem that it
> takes longer and longer for freeswitch to respond to DTMF input for some
> reason.
> Please refer to http://pastebin.freeswitch.org/16435 for debug logs.
> Notice that the difference in time between any adjacent pairs of the
> following is increasing as time goes by. Am I missing something here?
>        switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels
> 20ms
>        switch_rtp.c:3302 RTP RECV DTMF 2:960
> Thanks.
>
> Here's the javascript program:
> var dtmf_digits = "";
>
>
> function on_dtmf_28(session, type, digits, arg)
> {
>    if (type == "dtmf")
>    {
>
>        dtmf_digits = digits.digit;
>
>        console_log("dtmf_digits=" + dtmf_digits + "\n");
>
>        return false; // barge-in / done
>    }
>    return true;
> }
>
> while (true)
> {
>        dtmf_digits = "";
>        session.flushDigits();
>        session.streamFile("/usr/local/freeswitch/sounds/long-prompt.wav",
> on_dtmf_28, false);
>
>        if(dtmf_digits.length == 0)
>        {
>            /* no input */
>            console_log("no input\n");
>            session.speak('flite', 'kal', 'no input!', null);
>        }
>        else if(dtmf_digits == "1" || dtmf_digits == "2")
>        {
>                console_log("match "+dtmf_digits+"\n");
>                session.speak('flite', 'kal', 'you pressed ' + dtmf_digits+
> '!', null);
>        }
>        else
>        {
>            /* no match */
>            console_log("no match\n");
>            session.speak('flite', 'kal', "no match!", null);
>        }
>
> }
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Tue Jun  7 19:56:31 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 7 Jun 2011 16:56:31 +0100
Subject: [Freeswitch-users] A DTMF issue
In-Reply-To: <BANLkTikv5ULzkmnz86TT3zzANR1jb-Us5A@mail.gmail.com>
References: <33095823FD21DF429B481B5163264B7950AC585DEE@VMBX102.ihostexchange.net>
	<BANLkTi=vcvoLy=eU7f_=0bGCJ=2K2NjGfw@mail.gmail.com>
	<33095823FD21DF429B481B5163264B7950AC5861DC@VMBX102.ihostexchange.net>
	<BANLkTikv5ULzkmnz86TT3zzANR1jb-Us5A@mail.gmail.com>
Message-ID: <BANLkTikeWBGjGYAO7nBBJHhd7d1BPbySGA@mail.gmail.com>

- checks whether the session is still active (true anytime between call
starts and hangup)
- also session:ready will return false if the call is being
transferred. *Bottom line is you should always be checking
session:ready on any loops and
periodicly throughout your script and exit asap if it returns false.*

[http://wiki.freeswitch.org/wiki/Mod_lua#session:ready]



On 7 June 2011 16:54, Steven Ayre <steveayre at gmail.com> wrote:

> It's a simple fix. Always check session.ready when you loop. Yes, that's
> how it's designed to work.
>
> -Steve
>
>
>
> On 7 June 2011 16:05, Yungwei Chen <yungwei at resolvity.com> wrote:
>
>> That's how it's supposed to work and no workaround?
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
>> *Sent:* Friday, June 03, 2011 5:46 PM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] A DTMF issue
>>
>>
>>
>> You should be checking session.ready() in that loop, as it is I think
>> you'll find the scripts are never ending when the channel is hung up.
>>
>> streamFile/flushDigits probably return instantly, but then get called
>> again on the next loop iteration. The problem would get slightly worse after
>> each test call since there would be an extra copy of the script running.
>>
>> Try replacing:
>>     while(true) {
>> with:
>>     while(session.ready()) {
>>
>> -Steve
>>
>> On 3 June 2011 23:35, Yungwei Chen wrote:
>>
>> Hi,
>>
>> When testing the following javascript program, I notice a problem that it
>> takes longer and longer for freeswitch to respond to DTMF input for some
>> reason.
>> Please refer to http://pastebin.freeswitch.org/16435 for debug logs.
>> Notice that the difference in time between any adjacent pairs of the
>> following is increasing as time goes by. Am I missing something here?
>>        switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels
>> 20ms
>>        switch_rtp.c:3302 RTP RECV DTMF 2:960
>> Thanks.
>>
>> Here's the javascript program:
>> var dtmf_digits = "";
>>
>>
>> function on_dtmf_28(session, type, digits, arg)
>> {
>>    if (type == "dtmf")
>>    {
>>
>>        dtmf_digits = digits.digit;
>>
>>        console_log("dtmf_digits=" + dtmf_digits + "\n");
>>
>>        return false; // barge-in / done
>>    }
>>    return true;
>> }
>>
>> while (true)
>> {
>>        dtmf_digits = "";
>>        session.flushDigits();
>>        session.streamFile("/usr/local/freeswitch/sounds/long-prompt.wav",
>> on_dtmf_28, false);
>>
>>        if(dtmf_digits.length == 0)
>>        {
>>            /* no input */
>>            console_log("no input\n");
>>            session.speak('flite', 'kal', 'no input!', null);
>>        }
>>        else if(dtmf_digits == "1" || dtmf_digits == "2")
>>        {
>>                console_log("match "+dtmf_digits+"\n");
>>                session.speak('flite', 'kal', 'you pressed ' + dtmf_digits+
>> '!', null);
>>        }
>>        else
>>        {
>>            /* no match */
>>            console_log("no match\n");
>>            session.speak('flite', 'kal', "no match!", null);
>>        }
>>
>> }
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From awais-nazeer at hotmail.com  Tue Jun  7 20:05:28 2011
From: awais-nazeer at hotmail.com (awais nazir)
Date: Tue, 7 Jun 2011 22:05:28 +0600
Subject: [Freeswitch-users] Freeswitch dialplan help
Message-ID: <BAY156-w37D5149F53164FD75085D288630@phx.gbl>


Hi

I want to achieve a simple scenario , after attempting a bridge in my dialplan if I get cause code 480, it should be converted to 503 to originator.

<action application="bridge" data="sofia/${Dialstring}"/>
<!--If ${hangup_cause} == "480" then ${hangup_cause}
=503
>

Can somebody help here.

  		 	   		  
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From wstephen80 at gmail.com  Tue Jun  7 20:53:20 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 7 Jun 2011 18:53:20 +0200
Subject: [Freeswitch-users] Freeswitch dialplan help
In-Reply-To: <BAY156-w37D5149F53164FD75085D288630@phx.gbl>
References: <BAY156-w37D5149F53164FD75085D288630@phx.gbl>
Message-ID: <BANLkTi=mf3LjWiK=4o9TvUAK5bnKZ=RGEA@mail.gmail.com>

I think that something like this can work (not tested by me):

<context name="default">
    <extension name="myextension">
        <condition>
            <action application="bridge" data="sofia/${Dialstring}"/>
            <action application="transfer" data="dohangup XML dohangup"/>
        </condition>
    </extension>
</context>

<context name="dohangup">
    <extension name="dohangup">
        <condition field="${bridge_hangup_cause}" expression="^(*NO_ANSWER*)$"
break="on-true">
            <action application="hangup" data="*NORMAL_CIRCUIT_CONGESTION*
"/>
        </condition>
        <condition>
            <action application="hangup" data="${bridge_hangup_cause}"/>
        </condition>
    </extension>
</context>


Stephen

On Tue, Jun 7, 2011 at 6:05 PM, awais nazir <awais-nazeer at hotmail.com>wrote:

>  Hi
>
> I want to achieve a simple scenario , after attempting a bridge in my
> dialplan if I get cause code 480, it should be converted to 503 to
> originator.
>
> <action application="bridge" data="sofia/${Dialstring}"/>
> <!--If
>
> ${hangup_cause} == "480" then ${hangup_cause}
> =503
> >
>
> Can somebody help here.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From elijah at crankenstein.com  Tue Jun  7 21:05:10 2011
From: elijah at crankenstein.com (elijah)
Date: Tue, 7 Jun 2011 10:05:10 -0700
Subject: [Freeswitch-users] caller added to conference cannot be heard until
	members speak
Message-ID: <BANLkTikcFZ8Qak8s3S-ePmUEgamV2jnH8Q@mail.gmail.com>

Following the example in
http://wiki.freeswitch.org/wiki/Conference_Add_Call_Example I have
established a conference in which a moderator dials the number of a 3rd
member he wishes to conference in. Problem is, the newly added 3rd member
cannot be heard until after one of the original members speaks. This effect
causes the new member to be saying "hello" without anyone hearing. Can
you recommend a solution to get this 3rd member mixed into the conference
upon answer?
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From yungwei at resolvity.com  Tue Jun  7 21:06:15 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Tue, 7 Jun 2011 13:06:15 -0400
Subject: [Freeswitch-users] A DTMF issue
In-Reply-To: <BANLkTikpNteMa6oLw8ZCj=Z9q1XpcGYDkg@mail.gmail.com>
References: <33095823FD21DF429B481B5163264B7950AC585DEE@VMBX102.ihostexchange.net>
	<BANLkTi=vcvoLy=eU7f_=0bGCJ=2K2NjGfw@mail.gmail.com>
	<33095823FD21DF429B481B5163264B7950AC5861DC@VMBX102.ihostexchange.net>
	<BANLkTikpNteMa6oLw8ZCj=Z9q1XpcGYDkg@mail.gmail.com>
Message-ID: <33095823FD21DF429B481B5163264B7950AC5862B0@VMBX102.ihostexchange.net>

I tried session.ready(), and that doesn't fix the problem: response to DTMF input gets slower and slower as time goes by when one interacts with the javascript program in my original post.
Just to clarify, that's by design?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Tuesday, June 07, 2011 10:55 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] A DTMF issue

Yes, this is by design.

I think you might be trying to use the wrong tool for the task at hand. What is the problem that you are trying to solve? What function/feature are you trying to implement?

-MC
On Tue, Jun 7, 2011 at 8:05 AM, Yungwei Chen wrote:
That's how it's supposed to work and no workaround?

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Steven Ayre
Sent: Friday, June 03, 2011 5:46 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] A DTMF issue

You should be checking session.ready() in that loop, as it is I think you'll find the scripts are never ending when the channel is hung up.

streamFile/flushDigits probably return instantly, but then get called again on the next loop iteration. The problem would get slightly worse after each test call since there would be an extra copy of the script running.

Try replacing:
    while(true) {
with:
    while(session.ready()) {

-Steve
On 3 June 2011 23:35, Yungwei Chen wrote:
Hi,

When testing the following javascript program, I notice a problem that it takes longer and longer for freeswitch to respond to DTMF input for some reason.
Please refer to http://pastebin.freeswitch.org/16435 for debug logs.
Notice that the difference in time between any adjacent pairs of the following is increasing as time goes by. Am I missing something here?
       switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
       switch_rtp.c:3302 RTP RECV DTMF 2:960
Thanks.

Here's the javascript program:
var dtmf_digits = "";


function on_dtmf_28(session, type, digits, arg)
{
   if (type == "dtmf")
   {

       dtmf_digits = digits.digit;

       console_log("dtmf_digits=" + dtmf_digits + "\n");

       return false; // barge-in / done
   }
   return true;
}

while (true)
{
       dtmf_digits = "";
       session.flushDigits();
       session.streamFile("/usr/local/freeswitch/sounds/long-prompt.wav", on_dtmf_28, false);

       if(dtmf_digits.length == 0)
       {
           /* no input */
           console_log("no input\n");
           session.speak('flite', 'kal', 'no input!', null);
       }
       else if(dtmf_digits == "1" || dtmf_digits == "2")
       {
               console_log("match "+dtmf_digits+"\n");
               session.speak('flite', 'kal', 'you pressed ' + dtmf_digits+ '!', null);
       }
       else
       {
           /* no match */
           console_log("no match\n");
           session.speak('flite', 'kal', "no match!", null);
       }

}


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

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From chat2jesse at gmail.com  Tue Jun  7 21:26:14 2011
From: chat2jesse at gmail.com (jesse)
Date: Tue, 7 Jun 2011 10:26:14 -0700
Subject: [Freeswitch-users] Freeswitch 1.0.6 doesn't have mod_hash.c?
In-Reply-To: <570EC7DE-9D4A-48A4-8B8B-9F13FD51C033@gmail.com>
References: <4DED9F50.50109@tagnet.ru> <CA1320DB.170D0%krice@freeswitch.org>
	<BANLkTinfm+miJ2a=vNcMfVV1Hc3TAOcaEA@mail.gmail.com>
	<570EC7DE-9D4A-48A4-8B8B-9F13FD51C033@gmail.com>
Message-ID: <BANLkTincWoELr2VNrhSnx2rkchM40uhWNALxCTyM2XmKHf7iEg@mail.gmail.com>

Steve, thanks for your detailed elaboration, the error message disappeared
after loading mod_limit in 1.0.6. I will upgrade my system to 1.0.7 shortly.


On Tue, Jun 7, 2011 at 12:51 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Well more specifically, any module can register an application with any
> name it likes. In this case you're trying to call an application named hash
> which hasn't been loaded.
>
> In 1.0.6 that is registered by mod_limit, in more recent versions it's
> provided by the newer mod_hash which isn't in the older versions.
>
> Steve on iPhone
>
> On 7 Jun 2011, at 07:37, envelopes envelopes <sunwood360 at gmail.com> wrote:
>
> Sure, I am just curious about this error message since there is no
> mod_hash.c in 1.0.6.
> On Jun 6, 2011 10:29 PM, "Ken Rice" < <krice at freeswitch.org>
> krice at freeswitch.org> wrote:
> > How about quit using 1.0.6 and update to current...
> >
> >
> > On 6/6/11 10:47 PM, "Boris Kovalenko" < <boris at tagnet.ru>boris at tagnet.ru>
> wrote:
> >
> >> Hello!
> >>
> >> It is in source tree now. Please read Changelog
> >>
> >>
> >>> but my fs running time complains : switch_core_session.c:1731 Invalid
> >>> Application hash. I couldn't find mod_hash.c in src package...
> >>>
> >>>
> >>>
> >>> any idea?
> >>>
> >>>
> >>>
> >>>
> >>> thanks!
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> <FreeSWITCH-users at lists.freeswitch.org>
> FreeSWITCH-users at lists.freeswitch.org
> >>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> <http://www.freeswitch.org>http://www.freeswitch.org
> >>>
> >>
> >>
> >>
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Tue Jun  7 22:05:26 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 7 Jun 2011 19:05:26 +0100
Subject: [Freeswitch-users] A DTMF issue
In-Reply-To: <33095823FD21DF429B481B5163264B7950AC5862B0@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950AC585DEE@VMBX102.ihostexchange.net>
	<BANLkTi=vcvoLy=eU7f_=0bGCJ=2K2NjGfw@mail.gmail.com>
	<33095823FD21DF429B481B5163264B7950AC5861DC@VMBX102.ihostexchange.net>
	<BANLkTikpNteMa6oLw8ZCj=Z9q1XpcGYDkg@mail.gmail.com>
	<33095823FD21DF429B481B5163264B7950AC5862B0@VMBX102.ihostexchange.net>
Message-ID: <BANLkTing+4txvgHgP8OVgKv77LdB_9oHfQ@mail.gmail.com>

If session.ready isn't working then something else is causing the
slowness...

-Steve


On 7 June 2011 18:06, Yungwei Chen <yungwei at resolvity.com> wrote:

> I tried session.ready(), and that doesn't fix the problem: response to DTMF
> input gets slower and slower as time goes by when one interacts with the
> javascript program in my original post.
>
> Just to clarify, that's by design?
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Tuesday, June 07, 2011 10:55 AM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] A DTMF issue
>
>
>
> Yes, this is by design.
>
>
>
> I think you might be trying to use the wrong tool for the task at hand.
> What is the problem that you are trying to solve? What function/feature are
> you trying to implement?
>
>
>
> -MC
>
> On Tue, Jun 7, 2011 at 8:05 AM, Yungwei Chen wrote:
>
> That's how it's supposed to work and no workaround?
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* Friday, June 03, 2011 5:46 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] A DTMF issue
>
>
>
> You should be checking session.ready() in that loop, as it is I think
> you'll find the scripts are never ending when the channel is hung up.
>
> streamFile/flushDigits probably return instantly, but then get called again
> on the next loop iteration. The problem would get slightly worse after each
> test call since there would be an extra copy of the script running.
>
> Try replacing:
>     while(true) {
> with:
>     while(session.ready()) {
>
> -Steve
>
> On 3 June 2011 23:35, Yungwei Chen wrote:
>
> Hi,
>
> When testing the following javascript program, I notice a problem that it
> takes longer and longer for freeswitch to respond to DTMF input for some
> reason.
> Please refer to http://pastebin.freeswitch.org/16435 for debug logs.
> Notice that the difference in time between any adjacent pairs of the
> following is increasing as time goes by. Am I missing something here?
>        switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels
> 20ms
>        switch_rtp.c:3302 RTP RECV DTMF 2:960
> Thanks.
>
> Here's the javascript program:
> var dtmf_digits = "";
>
>
> function on_dtmf_28(session, type, digits, arg)
> {
>    if (type == "dtmf")
>    {
>
>        dtmf_digits = digits.digit;
>
>        console_log("dtmf_digits=" + dtmf_digits + "\n");
>
>        return false; // barge-in / done
>    }
>    return true;
> }
>
> while (true)
> {
>        dtmf_digits = "";
>        session.flushDigits();
>        session.streamFile("/usr/local/freeswitch/sounds/long-prompt.wav",
> on_dtmf_28, false);
>
>        if(dtmf_digits.length == 0)
>        {
>            /* no input */
>            console_log("no input\n");
>            session.speak('flite', 'kal', 'no input!', null);
>        }
>        else if(dtmf_digits == "1" || dtmf_digits == "2")
>        {
>                console_log("match "+dtmf_digits+"\n");
>                session.speak('flite', 'kal', 'you pressed ' + dtmf_digits+
> '!', null);
>        }
>        else
>        {
>            /* no match */
>            console_log("no match\n");
>            session.speak('flite', 'kal', "no match!", null);
>        }
>
> }
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Tue Jun  7 22:12:15 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 7 Jun 2011 19:12:15 +0100
Subject: [Freeswitch-users] Freeswitch dialplan help
In-Reply-To: <BANLkTi=mf3LjWiK=4o9TvUAK5bnKZ=RGEA@mail.gmail.com>
References: <BAY156-w37D5149F53164FD75085D288630@phx.gbl>
	<BANLkTi=mf3LjWiK=4o9TvUAK5bnKZ=RGEA@mail.gmail.com>
Message-ID: <BANLkTikxazXcx1t9xhtN1EOvUR01=E=KSw@mail.gmail.com>

The dohangup would probably be more like:

<context name="dohangup">
    <extension name="dohangup">
        <condition field="${bridge_hangup_cause}" expression="^(*NO_ANSWER*)$"
break="on-true">
            <action application="hangup" data="*NORMAL_CIRCUIT_CONGESTION*
"/>
            <anti-action application="hangup"
data="${bridge_hangup_cause}"/>
        </condition>
    </extension>
</context>

-Steve



On 7 June 2011 17:53, Stephen Wilde <wstephen80 at gmail.com> wrote:

> I think that something like this can work (not tested by me):
>
> <context name="default">
>     <extension name="myextension">
>         <condition>
>             <action application="bridge" data="sofia/${Dialstring}"/>
>             <action application="transfer" data="dohangup XML dohangup"/>
>          </condition>
>     </extension>
> </context>
>
> <context name="dohangup">
>     <extension name="dohangup">
>         <condition field="${bridge_hangup_cause}" expression="^(*NO_ANSWER
> *)$" break="on-true">
>             <action application="hangup" data="*NORMAL_CIRCUIT_CONGESTION*
> "/>
>         </condition>
>         <condition>
>             <action application="hangup" data="${bridge_hangup_cause}"/>
>         </condition>
>     </extension>
> </context>
>
>
> Stephen
>
> On Tue, Jun 7, 2011 at 6:05 PM, awais nazir <awais-nazeer at hotmail.com>wrote:
>
>>  Hi
>>
>> I want to achieve a simple scenario , after attempting a bridge in my
>> dialplan if I get cause code 480, it should be converted to 503 to
>> originator.
>>
>> <action application="bridge" data="sofia/${Dialstring}"/>
>> <!--If
>>
>> ${hangup_cause} == "480" then ${hangup_cause}
>> =503
>> >
>>
>> Can somebody help here.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kris at kriskinc.com  Tue Jun  7 22:35:03 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 7 Jun 2011 14:35:03 -0400
Subject: [Freeswitch-users] Freeswitch dialplan help
In-Reply-To: <BAY156-w37D5149F53164FD75085D288630@phx.gbl>
References: <BAY156-w37D5149F53164FD75085D288630@phx.gbl>
Message-ID: <BANLkTim0mfWDA9tONmSh02TbpL2L7UU85Q@mail.gmail.com>

Set continue_on_fail=true and use respond 503:

<action application="set" data="continue_on_fail=true"/>
<action application="bridge" data="blah"/>
<action application="respond" data="503"/>

On Tue, Jun 7, 2011 at 12:05 PM, awais nazir <awais-nazeer at hotmail.com> wrote:
> Hi
>
> I want to achieve a simple scenario , after attempting a bridge in my
> dialplan if I get cause code 480, it should be converted to 503 to
> originator.
>
> <action application="bridge" data="sofia/${Dialstring}"/>
> <!--If
>
> ${hangup_cause} == "480" then ${hangup_cause}
> =503
>>
>
> Can somebody help here.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner


From awais-nazeer at hotmail.com  Tue Jun  7 22:50:49 2011
From: awais-nazeer at hotmail.com (awais nazir)
Date: Wed, 8 Jun 2011 00:50:49 +0600
Subject: [Freeswitch-users] Freeswitch dialplan help
Message-ID: <BAY156-w57CD541515BAFB0E8CE55A88630@phx.gbl>


Thank you

Actually I don't want to respond 503 on all cases. I just want to mask response code 480 with 503. So that originator can never get 480.
Other cause codes should go naturally.

On Tue, Jun 7, 2011 at 11:35 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
Set continue_on_fail=true and use respond 503:



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

<action application="bridge" data="blah"/>

<action application="respond" data="503"/>



On Tue, Jun 7, 2011 at 12:05 PM, awais nazir <awais-nazeer at hotmail.com> wrote:

> Hi


 		 	   		  
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From kris at kriskinc.com  Tue Jun  7 22:56:51 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 7 Jun 2011 14:56:51 -0400
Subject: [Freeswitch-users] Freeswitch dialplan help
In-Reply-To: <BAY156-w57CD541515BAFB0E8CE55A88630@phx.gbl>
References: <BAY156-w57CD541515BAFB0E8CE55A88630@phx.gbl>
Message-ID: <BANLkTikJ0aRL=H0-ug_YWsie4uB=-EHMJw@mail.gmail.com>

Find the cause that 480 maps into a use that instead of true:

http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail

On Tue, Jun 7, 2011 at 2:50 PM, awais nazir <awais-nazeer at hotmail.com> wrote:
> Thank you
>
> Actually I don't want to respond 503 on all cases. I just want to mask
> response code 480 with 503. So that originator can never get 480.
> Other cause codes should go naturally.
>

-- 
Kristian Kielhofner


From maj at in-oren.ru  Tue Jun  7 23:17:59 2011
From: maj at in-oren.ru (Alexey E. Kashaev)
Date: Wed, 8 Jun 2011 01:17:59 +0600
Subject: [Freeswitch-users] how to use mod_mp4
In-Reply-To: <BANLkTikozhRaq-VgL+8Kt_K-tC_Th-QB3Q@mail.gmail.com>
References: <1306486049569-6410226.post@n2.nabble.com>
	<BANLkTi=4ky6Kj9gE94k_i7tDvdW+LcpjvQ@mail.gmail.com>
	<1306520931787-6412076.post@n2.nabble.com>
	<576485B0-53B9-4B03-A09E-8A6D76BA6CE5@gmail.com>
	<1306832036825-6421915.post@n2.nabble.com>
	<BANLkTikozhRaq-VgL+8Kt_K-tC_Th-QB3Q@mail.gmail.com>
Message-ID: <BANLkTikH3HVJBdZZs7Z35dpKRVRONRsf2g@mail.gmail.com>

2011/5/31 Sidharth Kshatriya <sid.kshatriya at gmail.com>:
> Try playing your mp4 in a player. There seems to be something wrong with
> it... or try another one -- follow the error message :-)

Hello,
I'm expecting that problem too. Tried a lot of videos, but all of them
can't be played successfully... Player plays these files as expected.
Can anyone share a "normal" test file? :)


From frankie.k.yiu at gmail.com  Tue Jun  7 23:24:41 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Tue, 7 Jun 2011 12:24:41 -0700
Subject: [Freeswitch-users] How to dial an extension in Freeswitch after the
	call in connected?
Message-ID: <BANLkTik45afNpPxRedjoC-oA4UuBtgcHDQ@mail.gmail.com>

Hi there,

I have a question regarding having FreeSwitch dials an extension number
after the call is connected.

Here is the condition:
Freeswitch would get a call information from database and make an outbound
call.  Once it is connected, and if there is an extension that we want to
reach at that number, we would wait 5 second and dial that extension
number.  Right now I am using C# code and I don't see any API would allow me
to dial an extension.

Could someone please help me?

Thanks,
Frankie
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From lists at telefaks.de  Tue Jun  7 23:36:53 2011
From: lists at telefaks.de (Peter Steinbach)
Date: Tue, 07 Jun 2011 21:36:53 +0200
Subject: [Freeswitch-users] Check if UA is still there?
Message-ID: <4DEE7DD5.4000900@telefaks.de>

Hello,

is there a way to check if a UA is still available? The reason is, that 
sometimes a channel hangs and is not being terminated because the UA was 
disconnected due to a network failure. As the media is outside 
Freeswitch (bypass media), it is difficult for us to determine, if the 
call is still active. Meanwhile the B leg to PSTN is also not hungup and 
produces costs.

Is there a way Freeswitch can detect this?  Or is there a way Freeswitch 
may send a SIP message to the UA by an external process and wait for an 
OK which can be captured e.g. by event socket interface?

An help is welcome.

-- 
With kind regards
Peter Steinbach

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de




From kris at kriskinc.com  Wed Jun  8 00:02:56 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 7 Jun 2011 16:02:56 -0400
Subject: [Freeswitch-users] Check if UA is still there?
In-Reply-To: <4DEE7DD5.4000900@telefaks.de>
References: <4DEE7DD5.4000900@telefaks.de>
Message-ID: <BANLkTimnRp55gv5BX0Mc8LbXyubdwwmF8w@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#unregister-on-options-fail
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#nat-options-ping

On Tue, Jun 7, 2011 at 3:36 PM, Peter Steinbach <lists at telefaks.de> wrote:
> Hello,
>
> is there a way to check if a UA is still available? The reason is, that
> sometimes a channel hangs and is not being terminated because the UA was
> disconnected due to a network failure. As the media is outside
> Freeswitch (bypass media), it is difficult for us to determine, if the
> call is still active. Meanwhile the B leg to PSTN is also not hungup and
> produces costs.
>
> Is there a way Freeswitch can detect this? ?Or is there a way Freeswitch
> may send a SIP message to the UA by an external process and wait for an
> OK which can be captured e.g. by event socket interface?
>
> An help is welcome.
>
> --
> With kind regards
> Peter Steinbach
>
> Telefaks Services GmbH
> mailto:lists (att) telefaks.de
> Internet: www.telefaks.de
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner


From elijah at crankenstein.com  Wed Jun  8 00:05:21 2011
From: elijah at crankenstein.com (elijah)
Date: Tue, 7 Jun 2011 13:05:21 -0700
Subject: [Freeswitch-users] caller added to conference cannot be heard
	until members speak
In-Reply-To: <BANLkTikcFZ8Qak8s3S-ePmUEgamV2jnH8Q@mail.gmail.com>
References: <BANLkTikcFZ8Qak8s3S-ePmUEgamV2jnH8Q@mail.gmail.com>
Message-ID: <BANLkTik_0pzD2gevJRYAoSWuyai+83F7qg@mail.gmail.com>

I suppose never mind. I fixed my issue by changing this:

<action application="set" data="res=${bgapi originate
{origination_uuid=${new_uuid}}sofia/gateway/onesource/$1
&conference(${conference_name})}"/>

to this:

<action application="set" data="res=${bgapi originate
{origination_uuid=${new_uuid},ignore_early_media=ring_ready,originate_timeout=180}sofia/gateway/onesource/$1
&conference(${conference_name})}"/>

I lose ring-tone on the outbound leg so if you have a more elegant solution
please let me know.


On Tue, Jun 7, 2011 at 10:05 AM, elijah <elijah at crankenstein.com> wrote:

> Following the example in
> http://wiki.freeswitch.org/wiki/Conference_Add_Call_Example I have
> established a conference in which a moderator dials the number of a 3rd
> member he wishes to conference in. Problem is, the newly added 3rd member
> cannot be heard until after one of the original members speaks. This effect
> causes the new member to be saying "hello" without anyone hearing. Can
> you recommend a solution to get this 3rd member mixed into the conference
> upon answer?
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From eric at loopfx.com  Wed Jun  8 00:16:07 2011
From: eric at loopfx.com (Eric Beard)
Date: Tue, 7 Jun 2011 16:16:07 -0400
Subject: [Freeswitch-users] SIP Response Headers
Message-ID: <D4A666272A214C4DBBABFE569CA34CA401DFF0331A@doctorbob.loop.local>

Hello,

I am trying to set SIP response headers using something like this in the dial plan:

<action application="set" data="sip_ph_X-MyVariable=true"/>

I haven't been able to get it into the SIP headers, however.

I have a simple setup for testing:

Soft Phone -> Freeswitch -> Gateway

I want Freeswitch to add the header to the SIP response sent back to the soft phone.

Any ideas what I might be missing?  I have also tried the sip_rh_ variant.

Thanks,

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com<mailto:eric at loopfx.com>

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From lists at telefaks.de  Wed Jun  8 00:46:55 2011
From: lists at telefaks.de (Peter Steinbach)
Date: Tue, 07 Jun 2011 22:46:55 +0200
Subject: [Freeswitch-users] Check if UA is still there?
In-Reply-To: <BANLkTimnRp55gv5BX0Mc8LbXyubdwwmF8w@mail.gmail.com>
References: <4DEE7DD5.4000900@telefaks.de>
	<BANLkTimnRp55gv5BX0Mc8LbXyubdwwmF8w@mail.gmail.com>
Message-ID: <4DEE8E3F.6050107@telefaks.de>

Thanks Kristian,

does that also mean that both legs of the call for this UA will be 
terminated on deregister?

Best regards
Peter

Am 07.06.2011 22:02, schrieb Kristian Kielhofner:
> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#unregister-on-options-fail
> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#nat-options-ping
>
> On Tue, Jun 7, 2011 at 3:36 PM, Peter Steinbach<lists at telefaks.de>  wrote:
>> Hello,
>>
>> is there a way to check if a UA is still available? The reason is, that
>> sometimes a channel hangs and is not being terminated because the UA was
>> disconnected due to a network failure. As the media is outside
>> Freeswitch (bypass media), it is difficult for us to determine, if the
>> call is still active. Meanwhile the B leg to PSTN is also not hungup and
>> produces costs.
>>
>> Is there a way Freeswitch can detect this?  Or is there a way Freeswitch
>> may send a SIP message to the UA by an external process and wait for an
>> OK which can be captured e.g. by event socket interface?
>>
>> An help is welcome.
>>
>> --
>> With kind regards
>> Peter Steinbach
>>
>> Telefaks Services GmbH
>> mailto:lists (att) telefaks.de
>> Internet: www.telefaks.de
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>


-- 
With kind regards
Peter Steinbach

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de




From brad at tech21.com  Wed Jun  8 00:51:29 2011
From: brad at tech21.com (Brad Mina)
Date: Tue, 7 Jun 2011 13:51:29 -0700
Subject: [Freeswitch-users] How to dial an extension in Freeswitch after
 the call in connected?
In-Reply-To: <BANLkTik45afNpPxRedjoC-oA4UuBtgcHDQ@mail.gmail.com>
References: <BANLkTik45afNpPxRedjoC-oA4UuBtgcHDQ@mail.gmail.com>
Message-ID: <BANLkTim_0_=fU2p9w2y9QNahJBvGmGcEog@mail.gmail.com>

Frankie,

Since the receiving end is looking to detect DTMF you might look into
calling a script to generate tones for the extensions in question.

http://wiki.freeswitch.org/wiki/TGML

On Tue, Jun 7, 2011 at 12:24 PM, Frankie Yiu <frankie.k.yiu at gmail.com>wrote:

> Hi there,
>
> I have a question regarding having FreeSwitch dials an extension number
> after the call is connected.
>
> Here is the condition:
> Freeswitch would get a call information from database and make an outbound
> call.  Once it is connected, and if there is an extension that we want to
> reach at that number, we would wait 5 second and dial that extension
> number.  Right now I am using C# code and I don't see any API would allow me
> to dial an extension.
>
> Could someone please help me?
>
> Thanks,
> Frankie
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From eric at loopfx.com  Wed Jun  8 00:51:34 2011
From: eric at loopfx.com (Eric Beard)
Date: Tue, 7 Jun 2011 16:51:34 -0400
Subject: [Freeswitch-users] Freeswitch dialplan help
In-Reply-To: <BANLkTi=mf3LjWiK=4o9TvUAK5bnKZ=RGEA@mail.gmail.com>
References: <BAY156-w37D5149F53164FD75085D288630@phx.gbl>
	<BANLkTi=mf3LjWiK=4o9TvUAK5bnKZ=RGEA@mail.gmail.com>
Message-ID: <D4A666272A214C4DBBABFE569CA34CA401DFF0338B@doctorbob.loop.local>

I tried this and it does not modify the SIP response code.  It seems to only affect this response header:

Reason: Q.850;cause=34;text="NORMAL_CIRCUIT_CONGESTION"

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com<mailto:eric at loopfx.com>

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stephen Wilde
Sent: Tuesday, June 07, 2011 12:53 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Freeswitch dialplan help

I think that something like this can work (not tested by me):

<context name="default">
    <extension name="myextension">
        <condition>
            <action application="bridge" data="sofia/${Dialstring}"/>
            <action application="transfer" data="dohangup XML dohangup"/>
        </condition>
    </extension>
</context>

<context name="dohangup">
    <extension name="dohangup">
        <condition field="${bridge_hangup_cause}" expression="^(NO_ANSWER)$" break="on-true">
            <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/>
        </condition>
        <condition>
            <action application="hangup" data="${bridge_hangup_cause}"/>
        </condition>
    </extension>
</context>


Stephen

On Tue, Jun 7, 2011 at 6:05 PM, awais nazir <awais-nazeer at hotmail.com<mailto:awais-nazeer at hotmail.com>> wrote:
Hi

I want to achieve a simple scenario , after attempting a bridge in my dialplan if I get cause code 480, it should be converted to 503 to originator.

<action application="bridge" data="sofia/${Dialstring}"/>
<!--If

${hangup_cause} == "480" then ${hangup_cause}

=503
>

Can somebody help here.



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

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From kris at kriskinc.com  Wed Jun  8 01:05:57 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 7 Jun 2011 17:05:57 -0400
Subject: [Freeswitch-users] Freeswitch dialplan help
In-Reply-To: <D4A666272A214C4DBBABFE569CA34CA401DFF0338B@doctorbob.loop.local>
References: <BAY156-w37D5149F53164FD75085D288630@phx.gbl>
	<BANLkTi=mf3LjWiK=4o9TvUAK5bnKZ=RGEA@mail.gmail.com>
	<D4A666272A214C4DBBABFE569CA34CA401DFF0338B@doctorbob.loop.local>
Message-ID: <BANLkTikPXqsZnK4VWOdCmV9d2=ugHCmJ6g@mail.gmail.com>

I process over one million calls per day with this exact
configuration.  I'm certain it works and does modify the SIP response
code.  My peer in this configuration depends on getting a 503.

On Tue, Jun 7, 2011 at 4:51 PM, Eric Beard <eric at loopfx.com> wrote:
> I tried this and it does not modify the SIP response code.? It seems to only
> affect this response header:
>
>
>
> Reason: Q.850;cause=34;text="NORMAL_CIRCUIT_CONGESTION"
>
>
>
> -----------------------
>
> Eric Z. Beard, CTO
>
> Loop LLC
>
> w (877) 850-2010 x9249
>
> m (727) 776-2768
>
> eric at loopfx.com
>
>

-- 
Kristian Kielhofner


From kris at kriskinc.com  Wed Jun  8 01:07:58 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 7 Jun 2011 17:07:58 -0400
Subject: [Freeswitch-users] Check if UA is still there?
In-Reply-To: <4DEE8E3F.6050107@telefaks.de>
References: <4DEE7DD5.4000900@telefaks.de>
	<BANLkTimnRp55gv5BX0Mc8LbXyubdwwmF8w@mail.gmail.com>
	<4DEE8E3F.6050107@telefaks.de>
Message-ID: <BANLkTinOqhxOD5LzmRkvDrek4g1J+K3i_Q@mail.gmail.com>

Peter,

  No.  Registration is separate from any calls.  If the device fails
to respond to OPTIONS it will be de-registered and inbound calls to
that user/device will fail immediately because it isn't registered.
Any ongoing calls with or incoming calls from that device will proceed
as normal.

On Tue, Jun 7, 2011 at 4:46 PM, Peter Steinbach <lists at telefaks.de> wrote:
> Thanks Kristian,
>
> does that also mean that both legs of the call for this UA will be
> terminated on deregister?
>
> Best regards
> Peter

-- 
Kristian Kielhofner


From steveayre at gmail.com  Wed Jun  8 02:55:43 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 7 Jun 2011 23:55:43 +0100
Subject: [Freeswitch-users] SIP Response Headers
In-Reply-To: <D4A666272A214C4DBBABFE569CA34CA401DFF0331A@doctorbob.loop.local>
References: <D4A666272A214C4DBBABFE569CA34CA401DFF0331A@doctorbob.loop.local>
Message-ID: <BANLkTikMmPsN_PvO7BQF0VWNhKsMkEZLAw@mail.gmail.com>

What about the sip_h_X- variant?

It depends which you want it to go into. sip_ph_X will only show in
100/180/183 provisional responses.


-Steve




On 7 June 2011 21:16, Eric Beard <eric at loopfx.com> wrote:

> Hello,
>
>
>
> I am trying to set SIP response headers using something like this in the
> dial plan:
>
>
>
> <action application="set" data="sip_ph_X-MyVariable=true"/>
>
>
>
> I haven?t been able to get it into the SIP headers, however.
>
>
>
> I have a simple setup for testing:
>
>
>
> Soft Phone -> Freeswitch -> Gateway
>
>
>
> I want Freeswitch to add the header to the SIP response sent back to the
> soft phone.
>
>
>
> Any ideas what I might be missing?  I have also tried the sip_rh_ variant.
>
>
>
> Thanks,
>
>
>
> -----------------------
>
> *Eric Z. Beard, CTO*
>
> Loop LLC
>
> w (877) 850-2010 x9249
>
> m (727) 776-2768
>
> eric at loopfx.com
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Wed Jun  8 03:00:23 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 8 Jun 2011 00:00:23 +0100
Subject: [Freeswitch-users] How to dial an extension in Freeswitch after
 the call in connected?
In-Reply-To: <BANLkTim_0_=fU2p9w2y9QNahJBvGmGcEog@mail.gmail.com>
References: <BANLkTik45afNpPxRedjoC-oA4UuBtgcHDQ@mail.gmail.com>
	<BANLkTim_0_=fU2p9w2y9QNahJBvGmGcEog@mail.gmail.com>
Message-ID: <BANLkTi=oxScEPcnS6hXViTjFVL+Siu60iQ@mail.gmail.com>

Or use the dialplan apps that already exist for that purpose, and will
automatically choose whether to send it inband or out-of-band:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_queue_dtmf
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_send_dtmf

For example:
<action application="queue_dtmf" data="WWWWW1234"/>
<action application="bridge" data="sofia/gateway/gw1/555"/>

This would dial 555 on gw1, and when answered wait 5s then dial 1234. W is a
1 second delay, so WWWWW is a 5 second delay.

-Steve



On 7 June 2011 21:51, Brad Mina <brad at tech21.com> wrote:

> Frankie,
>
> Since the receiving end is looking to detect DTMF you might look into
> calling a script to generate tones for the extensions in question.
>
> http://wiki.freeswitch.org/wiki/TGML
>
> On Tue, Jun 7, 2011 at 12:24 PM, Frankie Yiu <frankie.k.yiu at gmail.com>wrote:
>
>> Hi there,
>>
>> I have a question regarding having FreeSwitch dials an extension number
>> after the call is connected.
>>
>> Here is the condition:
>> Freeswitch would get a call information from database and make an outbound
>> call.  Once it is connected, and if there is an extension that we want to
>> reach at that number, we would wait 5 second and dial that extension
>> number.  Right now I am using C# code and I don't see any API would allow me
>> to dial an extension.
>>
>> Could someone please help me?
>>
>> Thanks,
>> Frankie
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Wed Jun  8 03:02:19 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 8 Jun 2011 00:02:19 +0100
Subject: [Freeswitch-users] How to dial an extension in Freeswitch after
 the call in connected?
In-Reply-To: <BANLkTi=oxScEPcnS6hXViTjFVL+Siu60iQ@mail.gmail.com>
References: <BANLkTik45afNpPxRedjoC-oA4UuBtgcHDQ@mail.gmail.com>
	<BANLkTim_0_=fU2p9w2y9QNahJBvGmGcEog@mail.gmail.com>
	<BANLkTi=oxScEPcnS6hXViTjFVL+Siu60iQ@mail.gmail.com>
Message-ID: <BANLkTik+ez4cVFSteBaX=jUM1JCtmuiByA@mail.gmail.com>

There's also a api app so you can do it from ESL:
http://wiki.freeswitch.org/wiki/Mod_commands#uuid_send_dtmf

-Steve



On 8 June 2011 00:00, Steven Ayre <steveayre at gmail.com> wrote:

> Or use the dialplan apps that already exist for that purpose, and will
> automatically choose whether to send it inband or out-of-band:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_queue_dtmf
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_send_dtmf
>
> For example:
> <action application="queue_dtmf" data="WWWWW1234"/>
> <action application="bridge" data="sofia/gateway/gw1/555"/>
>
> This would dial 555 on gw1, and when answered wait 5s then dial 1234. W is
> a 1 second delay, so WWWWW is a 5 second delay.
>
> -Steve
>
>
>
>
> On 7 June 2011 21:51, Brad Mina <brad at tech21.com> wrote:
>
>> Frankie,
>>
>> Since the receiving end is looking to detect DTMF you might look into
>> calling a script to generate tones for the extensions in question.
>>
>> http://wiki.freeswitch.org/wiki/TGML
>>
>> On Tue, Jun 7, 2011 at 12:24 PM, Frankie Yiu <frankie.k.yiu at gmail.com>wrote:
>>
>>> Hi there,
>>>
>>> I have a question regarding having FreeSwitch dials an extension number
>>> after the call is connected.
>>>
>>> Here is the condition:
>>> Freeswitch would get a call information from database and make an
>>> outbound call.  Once it is connected, and if there is an extension that we
>>> want to reach at that number, we would wait 5 second and dial that extension
>>> number.  Right now I am using C# code and I don't see any API would allow me
>>> to dial an extension.
>>>
>>> Could someone please help me?
>>>
>>> Thanks,
>>> Frankie
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From steveayre at gmail.com  Wed Jun  8 03:03:39 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 8 Jun 2011 00:03:39 +0100
Subject: [Freeswitch-users] Freeswitch dialplan help
In-Reply-To: <D4A666272A214C4DBBABFE569CA34CA401DFF0338B@doctorbob.loop.local>
References: <BAY156-w37D5149F53164FD75085D288630@phx.gbl>
	<BANLkTi=mf3LjWiK=4o9TvUAK5bnKZ=RGEA@mail.gmail.com>
	<D4A666272A214C4DBBABFE569CA34CA401DFF0338B@doctorbob.loop.local>
Message-ID: <BANLkTimyXQHktQuCh3RPTPaUWDw_EaWWTg@mail.gmail.com>

Try also setting this before the bridge:
<action application="set" data="sip_ignore_remote_cause=true"/>

Without that you'll get a 34 reason but 480 sip code, with it you'll get a
34+503.

-Steve



On 7 June 2011 21:51, Eric Beard <eric at loopfx.com> wrote:

> I tried this and it does not modify the SIP response code.  It seems to
> only affect this response header:
>
>
>
> Reason: Q.850;cause=34;text="NORMAL_CIRCUIT_CONGESTION"
>
>
>
> -----------------------
>
> *Eric Z. Beard, CTO*
>
> Loop LLC
>
> w (877) 850-2010 x9249
>
> m (727) 776-2768
>
> eric at loopfx.com
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Stephen
> Wilde
> *Sent:* Tuesday, June 07, 2011 12:53 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Freeswitch dialplan help
>
>
>
> I think that something like this can work (not tested by me):
>
>
>
> <context name="default">
>
>     <extension name="myextension">
>
>         <condition>
>
>             <action application="bridge" data="sofia/${Dialstring}"/>
>
>             <action application="transfer" data="dohangup XML dohangup"/>
>
>         </condition>
>
>     </extension>
>
> </context>
>
>
>
> <context name="dohangup">
>
>     <extension name="dohangup">
>
>         <condition field="${bridge_hangup_cause}" expression="^(*NO_ANSWER
> *)$" break="on-true">
>
>             <action application="hangup" data="*NORMAL_CIRCUIT_CONGESTION*
> "/>
>
>         </condition>
>
>         <condition>
>
>             <action application="hangup" data="${bridge_hangup_cause}"/>
>
>         </condition>
>
>     </extension>
>
> </context>
>
>
>
>
>
> Stephen
>
>
>
> On Tue, Jun 7, 2011 at 6:05 PM, awais nazir <awais-nazeer at hotmail.com>
> wrote:
>
> Hi
>
> I want to achieve a simple scenario , after attempting a bridge in my
> dialplan if I get cause code 480, it should be converted to 503 to
> originator.
>
> <action application="bridge" data="sofia/${Dialstring}"/>
> <!--If
>
> ${hangup_cause} == "480" then ${hangup_cause}
>
> =503
> >
>
> Can somebody help here.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Jun  8 03:46:00 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 7 Jun 2011 16:46:00 -0700
Subject: [Freeswitch-users] Freeswitch dialplan help
In-Reply-To: <BANLkTimyXQHktQuCh3RPTPaUWDw_EaWWTg@mail.gmail.com>
References: <BAY156-w37D5149F53164FD75085D288630@phx.gbl>
	<BANLkTi=mf3LjWiK=4o9TvUAK5bnKZ=RGEA@mail.gmail.com>
	<D4A666272A214C4DBBABFE569CA34CA401DFF0338B@doctorbob.loop.local>
	<BANLkTimyXQHktQuCh3RPTPaUWDw_EaWWTg@mail.gmail.com>
Message-ID: <BANLkTi=ufmdiP0_YBUA7bokmooyS+ScL2g@mail.gmail.com>

Thanks for the tip Steve!
-MC

On Tue, Jun 7, 2011 at 4:03 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Try also setting this before the bridge:
> <action application="set" data="sip_ignore_remote_cause=true"/>
>
> Without that you'll get a 34 reason but 480 sip code, with it you'll get a
> 34+503.
>
> -Steve
>
>
>
>
> On 7 June 2011 21:51, Eric Beard <eric at loopfx.com> wrote:
>
>> I tried this and it does not modify the SIP response code.  It seems to
>> only affect this response header:
>>
>>
>>
>> Reason: Q.850;cause=34;text="NORMAL_CIRCUIT_CONGESTION"
>>
>>
>>
>> -----------------------
>>
>> *Eric Z. Beard, CTO*
>>
>> Loop LLC
>>
>> w (877) 850-2010 x9249
>>
>> m (727) 776-2768
>>
>> eric at loopfx.com
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Stephen
>> Wilde
>> *Sent:* Tuesday, June 07, 2011 12:53 PM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] Freeswitch dialplan help
>>
>>
>>
>> I think that something like this can work (not tested by me):
>>
>>
>>
>> <context name="default">
>>
>>     <extension name="myextension">
>>
>>         <condition>
>>
>>             <action application="bridge" data="sofia/${Dialstring}"/>
>>
>>             <action application="transfer" data="dohangup XML dohangup"/>
>>
>>         </condition>
>>
>>     </extension>
>>
>> </context>
>>
>>
>>
>> <context name="dohangup">
>>
>>     <extension name="dohangup">
>>
>>         <condition field="${bridge_hangup_cause}" expression="^(*
>> NO_ANSWER*)$" break="on-true">
>>
>>             <action application="hangup" data="*NORMAL_CIRCUIT_CONGESTION
>> *"/>
>>
>>         </condition>
>>
>>         <condition>
>>
>>             <action application="hangup" data="${bridge_hangup_cause}"/>
>>
>>         </condition>
>>
>>     </extension>
>>
>> </context>
>>
>>
>>
>>
>>
>> Stephen
>>
>>
>>
>> On Tue, Jun 7, 2011 at 6:05 PM, awais nazir <awais-nazeer at hotmail.com>
>> wrote:
>>
>> Hi
>>
>> I want to achieve a simple scenario , after attempting a bridge in my
>> dialplan if I get cause code 480, it should be converted to 503 to
>> originator.
>>
>> <action application="bridge" data="sofia/${Dialstring}"/>
>> <!--If
>>
>> ${hangup_cause} == "480" then ${hangup_cause}
>>
>> =503
>> >
>>
>> Can somebody help here.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cyril.zlachevsky at gmail.com  Wed Jun  8 04:16:45 2011
From: cyril.zlachevsky at gmail.com (Cyril Zlachevsky)
Date: Wed, 08 Jun 2011 03:16:45 +0300
Subject: [Freeswitch-users] ACL
Message-ID: <4DEEBF6D.7050608@gmail.com>

Hi,
Can't stand what should I do for ACL working.

I'm forwarding a call from a particular static IP to number at voipprovider.
I refused of using FS default configs because they are complicated and redundant for me.

This is my configuration:
<?xml version="1.0"?>
<document type="freeswitch/xml">

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

<X-PRE-PROCESS cmd="set" data="sipnet_proxy=voiprovider.com"/>
<X-PRE-PROCESS cmd="set" data="sipnet_login=echo"/>
<X-PRE-PROCESS cmd="set" data="sipnet_password=echo"/>

<X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/>

<section name="configuration">

<configuration name="switch.conf">
   <settings>
     <param name="colorize-console" value="true"/>
     <param name="max-sessions" value="1000"/>
     <param name="sessions-per-second" value="30"/>
     <param name="loglevel" value="debug"/>
   </settings>
</configuration>

<configuration name="modules.conf">
   <modules>
     <load module="mod_console"/>
     <load module="mod_logfile"/>
     <load module="mod_sofia"/>
     <load module="mod_dialplan_xml"/>
     <load module="mod_dptools"/>
     <load module="mod_commands"/>
     <load module="mod_expr"/>
     <load module="mod_sndfile"/>
   </modules>
</configuration>

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>

<configuration name="console.conf">
   <mappings>
     <map name="all" value="console,debug,info,notice,warning,err,crit,alert"/>
   </mappings>
   <settings>
     <param name="colorize" value="true"/>
     <param name="loglevel" value="info"/>
   </settings>
</configuration>

<configuration name="logfile.conf">
   <settings>
     <param name="rotate-on-hup" value="true"/>
   </settings>
   <profiles>
     <profile name="default">
       <settings>
         <param name="logfile" value="/var/log/freeswitch/freeswitch.log"/>
         <param name="rollover" value="10485760"/>
       </settings>
       <mappings>
         <map name="all" value="debug,info,notice,warning,err,crit,alert"/>
       </mappings>
     </profile>
   </profiles>
</configuration>

<configuration name="acl.conf" description="Network Lists">
   <network-lists>
     <list name="strict" default="allow">
       <node type="allow" cidr="195.225.XXX.XXX/32"/>
     </list>
     <list name="domains" default="deny">
       <node type="allow" domain="$${domain}"/>
     </list>
   </network-lists>
</configuration>

<configuration name="sofia.conf">
   <global_settings>
     <param name="log-level" value="0"/>
     <param name="debug-presence" value="0"/>
   </global_settings>
   <profiles>
     <profile name="local">
       <domains>
         <domain name="all" alias="true" parse="false"/>
       </domains>
       <settings>
         <param name="context" value="local"/>
         <param name="sip-port" value="5060"/>
         <param name="dialplan" value="XML"/>
         <param name="rtp-ip" value="$${local_address}"/>
         <param name="sip-ip" value="$${local_address}"/>
       </settings>
     </profile>
     <profile name="sipnet">
       <domains>
         <domain name="all" alias="false" parse="false"/>
       </domains>
       <settings>
         <param name="context" value="sipnet"/>
         <param name="sip-port" value="5080"/>
         <param name="dialplan" value="XML"/>
         <param name="rtp-ip" value="$${local_address}"/>
         <param name="sip-ip" value="$${local_address}"/>
         <!--param name="auth-calls" value="false"/-->
         <param name="auth-calls" value="true"/>
         <param name="apply-inbound-acl" value="strict"/>
       </settings>
       <gateways>
         <gateway name="sipnet">
           <param name="proxy" value="$${sipnet_proxy}"/>
           <param name="realm" value="asterisk"/>
           <param name="username" value="$${sipnet_login}"/>
           <param name="password" value="$${sipnet_password}"/>
           <param name="local-network-acl" value="strict"/>
         </gateway>
       </gateways>
     </profile>
   </profiles>
  </configuration>
</section>

<section name="dialplan">
   <context name="local">
     <extension name="local-accounts">
       <condition field="destination_number" expression="^(10[1-3])$">
         <action application="bridge" data="user/$1@$${local_address}"/>
       </condition>
     </extension>
     <extension name="sipnet">
       <condition field="destination_number" expression="^(\d+)$">
         <action application="set" data="effective_caller_id_number=$${sipnet_login}"/>
         <action application="bridge"
data="{sip_invite_domain=$${sipnet_proxy}}sofia/sipnet/$1@$${sipnet_proxy}"/>
       </condition>
     </extension>
   </context>
</section>

<section name="directory">
   <domain name="$${local_address}">
     <params>
       <param name="dial-string"
value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
     </params>
     <groups>
       <group name="local">
         <users>
           <user id="inboundtest"></user>
           <params>
             <param name="debug" value="yes"/>
             <param name="register" value="true"/>
             <param name="apply-inbound-acl" value="strict"/>
             <param name="apply-register-acl" value="strict"/>
           </params>
          </users>
       </group>
     </groups>
   </domain>
</section>
</document>

When I start FS, I can't see my IP 195.225.XXX.XXX in freeswitch.log - only this:
[NOTICE] switch_core.c:1088 Created ip list rfc1918.auto default (deny)
[NOTICE] switch_utils.c:248 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto
[NOTICE] switch_utils.c:248 Adding 172.16.0.0/12 (allow) [] to list rfc1918.auto
[NOTICE] switch_utils.c:248 Adding 192.168.0.0/16 (allow) [] to list rfc1918.auto
[NOTICE] switch_core.c:1096 Created ip list wan.auto default (allow)
[NOTICE] switch_utils.c:248 Adding 10.0.0.0/8 (deny) [] to list wan.auto
[NOTICE] switch_utils.c:248 Adding 172.16.0.0/12 (deny) [] to list wan.auto
[NOTICE] switch_utils.c:248 Adding 192.168.0.0/16 (deny) [] to list wan.auto
[NOTICE] switch_core.c:1104 Created ip list nat.auto default (deny)
[NOTICE] switch_core.c:1106 Adding 88.198.XXX.XXX/255.255.255.255 (deny) to list nat.auto
[NOTICE] switch_utils.c:248 Adding 10.0.0.0/8 (allow) [] to list nat.auto
[NOTICE] switch_utils.c:248 Adding 172.16.0.0/12 (allow) [] to list nat.auto
[NOTICE] switch_utils.c:248 Adding 192.168.0.0/16 (allow) [] to list nat.auto
[NOTICE] switch_core.c:1115 Created ip list loopback.auto default (deny)
[NOTICE] switch_utils.c:248 Adding 127.0.0.0/8 (allow) [] to list loopback.auto
[NOTICE] switch_core.c:1121 Created ip list localnet.auto default (deny)
[NOTICE] switch_core.c:1124 Adding 88.198.XXX.XXX/255.255.255.255 (allow) to list localnet.auto

With my current configuration FS allow to register from any IP.
Where is my error?


From mtaylor at employees.org  Wed Jun  8 05:46:21 2011
From: mtaylor at employees.org (Mike Taylor)
Date: Wed, 08 Jun 2011 13:46:21 +1200
Subject: [Freeswitch-users] Conceptual/design question - per gateway/DTMF &
	PSQL
In-Reply-To: <CA152D11.2FFDF%mike.taylor@staff.snap.net.nz>
Message-ID: <CA152D95.2FFE8%mike.taylor@staff.snap.net.nz>

Hi,

I hope that I've taught myself enough over the last few weeks that this
isn't a blindingly stupid question?

I am providing services to multiple customers on FS 1.0.6

I have a basic setup; (which I have inherited, hence the basic questions);

AS5350--------<external>----- FS-----<internal>------customer GW(s)
(No Registration) (No Registration)

All of the GW info is pulled from a PSWL database. Carriers, carrier_id, lcr
table etc.

The bit I am missing is;

If all of my GW config is in the PSQL database, and that's all accessed by
mod_lcr before routing the call, how do I allow for different DTMF types,
codecs etc, PER CUSTOMER GW ?

Am I stuck with actually defining the gateways statically under
/etc/freeswitch/conf/sip_profiles/ (or I see I can doit in the dialplan, but
that won't achieve what I need)

I was hoping to be able to pull the DTMF (and other misc info) from the
database, but the more I look into it,  the less likely it is that this will
work I think. (mod_lcr is already at the routing stage, can't modify DTMF at
that point AFAIK)

Like I said, hope the question isn't so obvious that it's crazy?

Regards,
Mike


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From p.devarahalli at alcatel-lucent.com  Wed Jun  8 08:03:32 2011
From: p.devarahalli at alcatel-lucent.com (DEVARAHALLI, PRAKASH (PRAKASH))
Date: Wed, 8 Jun 2011 09:33:32 +0530
Subject: [Freeswitch-users] FS does not uses the IP/PORT resolved
	using	STUN.
In-Reply-To: <BANLkTikmdM4qQ7x9Z753KenzxwmQwTFbDw@mail.gmail.com>
References: <7E79416ABDDDFC4E8C85571D0B5CAB3F138AE814F1@INBANSXCHMBSA1.in.alcatel-lucent.com>
	<BANLkTikmdM4qQ7x9Z753KenzxwmQwTFbDw@mail.gmail.com>
Message-ID: <7E79416ABDDDFC4E8C85571D0B5CAB3F138AE815F5@INBANSXCHMBSA1.in.alcatel-lucent.com>

Hi,
	Thanks for the response. I agree that the solution proposed by you works perfectly. But purpose of my test is to specifically use the STUN for NAT traversing.  

Thanks
Prakash

-----Original Message-----
From: guru singh [mailto:grsingh750 at gmail.com] 
Sent: Tuesday, June 07, 2011 2:44 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FS does not uses the IP/PORT resolved using STUN.

Hi,

I've never used STUN but if you have a router that supports UPnP/PMP,
you could try configuring your sofia profile to use auto-nat , it
usually does the job.

Regards,
guru

On Tue, Jun 7, 2011 at 1:23 PM, DEVARAHALLI, PRAKASH (PRAKASH)
<p.devarahalli at alcatel-lucent.com> wrote:
> I configured FS to use the STUN server to find the public IP address used by
> the NAT.
> FS sends the request to STUN and receives the reply from the STUN Server ,
> But it does
> not uses this info ( IP and port obtained from STUN)? neither while
> negotiation ( in SDP)
> nor in the RTP.
>
>
> My testbed:
> -----------
>
> < FreeSwitch > ---- < NAT > ---+----<NAT>--- < SIP Phone Client >
> ???????????????????????????????????????????? |
> ???????????????????????????????????????????? |
> ??????????????????????????????? [ STUN Server]
>
>
> File conf/sip_profiles/internal.xml has below configuration.
>
> ?? <param name="ext-rtp-ip" value="stun:120.120.120.120"/> ; This is the
> local STUN server.
> ?? <param name="ext-sip-ip" value="stun:120.120.120.120"/>
> ?? <param name="stun-auto-disable" value="false"/>
> ?? <param name="rtp-ip" value="auto"/>
>
> Thanks
> Prakash
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>




From david.ponzone at ipeva.fr  Wed Jun  8 10:04:41 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 8 Jun 2011 08:04:41 +0200
Subject: [Freeswitch-users] Freeswitch dialplan help
In-Reply-To: <BANLkTi=ufmdiP0_YBUA7bokmooyS+ScL2g@mail.gmail.com>
References: <BAY156-w37D5149F53164FD75085D288630@phx.gbl>
	<BANLkTi=mf3LjWiK=4o9TvUAK5bnKZ=RGEA@mail.gmail.com>
	<D4A666272A214C4DBBABFE569CA34CA401DFF0338B@doctorbob.loop.local>
	<BANLkTimyXQHktQuCh3RPTPaUWDw_EaWWTg@mail.gmail.com>
	<BANLkTi=ufmdiP0_YBUA7bokmooyS+ScL2g@mail.gmail.com>
Message-ID: <5990920F-0369-4EC3-84A0-7023FE9EEC99@ipeva.fr>

Personally, I used to do it this way:

    <extension name="from_gw_to_internal">
      <condition field="destination_number" expression="^(.*)$">
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=19"/>
        <action application="bridge" data="{sip_cid_type=none}sofia/gateway/gw/$1"/>
        <action application="transfer" data="480to503"/>
      </condition>
    </extension>

    <extension name="480to503">
      <condition field="${proto_specific_hangup_cause}" expression="sip:480">
        <action application="set" data="sip_ignore_remote_cause=true"/>
        <action application="respond" data="503"/>
        <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/>
      </condition>
    </extension>

Of course, you would have to change continue_on_fail=19 to whatever you want, as there are several possibilities for 480.

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

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

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




Le 08/06/2011 ? 01:46, Michael Collins a ?crit :

> Thanks for the tip Steve!
> -MC
> 
> On Tue, Jun 7, 2011 at 4:03 PM, Steven Ayre <steveayre at gmail.com> wrote:
> Try also setting this before the bridge:
> <action application="set" data="sip_ignore_remote_cause=true"/>
> 
> Without that you'll get a 34 reason but 480 sip code, with it you'll get a 34+503.
> 
> -Steve
> 
> 
> 
> 
> On 7 June 2011 21:51, Eric Beard <eric at loopfx.com> wrote:
> I tried this and it does not modify the SIP response code.  It seems to only affect this response header:
> 
>  
> Reason: Q.850;cause=34;text="NORMAL_CIRCUIT_CONGESTION"
> 
>  
> -----------------------
> 
> Eric Z. Beard, CTO
> 
> Loop LLC
> 
> w (877) 850-2010 x9249
> 
> m (727) 776-2768
> 
> eric at loopfx.com
> 
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stephen Wilde
> Sent: Tuesday, June 07, 2011 12:53 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Freeswitch dialplan help
> 
>  
> I think that something like this can work (not tested by me):
> 
>  
> <context name="default">
> 
>     <extension name="myextension">
> 
>         <condition>
> 
>             <action application="bridge" data="sofia/${Dialstring}"/>
> 
>             <action application="transfer" data="dohangup XML dohangup"/>
> 
>         </condition>
> 
>     </extension>
> 
> </context>
> 
>  
> <context name="dohangup">
> 
>     <extension name="dohangup">
> 
>         <condition field="${bridge_hangup_cause}" expression="^(NO_ANSWER)$" break="on-true">
> 
>             <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/>
> 
>         </condition>
> 
>         <condition>
> 
>             <action application="hangup" data="${bridge_hangup_cause}"/>
> 
>         </condition>
> 
>     </extension>
> 
> </context>
> 
>  
>  
> Stephen
> 
>  
> On Tue, Jun 7, 2011 at 6:05 PM, awais nazir <awais-nazeer at hotmail.com> wrote:
> 
> Hi
> 
> I want to achieve a simple scenario , after attempting a bridge in my dialplan if I get cause code 480, it should be converted to 503 to originator.
> 
> <action application="bridge" data="sofia/${Dialstring}"/>
> <!--If 
> 
> ${hangup_cause} == "480" then ${hangup_cause}
> =503
> >
> 
> Can somebody help here.
>  
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
>  
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at ipeva.fr  Wed Jun  8 10:08:09 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 8 Jun 2011 08:08:09 +0200
Subject: [Freeswitch-users] Conceptual/design question - per
	gateway/DTMF & PSQL
In-Reply-To: <CA152D95.2FFE8%mike.taylor@staff.snap.net.nz>
References: <CA152D95.2FFE8%mike.taylor@staff.snap.net.nz>
Message-ID: <00014BE5-06FB-4482-ADD1-0D5F44B25BF0@ipeva.fr>

Depending on the number of customers you intend to have, you can have one SIP profile per customer.
But that defeats slightly the purpose of your PSQL DB.

I don't know much mod_lcr, but if it can send back specific parameters for a gateway (codec_string, etc...), it can probably achieve what you need, but you may have to enable late legotiation.

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

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

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




Le 08/06/2011 ? 03:46, Mike Taylor a ?crit :

> Hi,
> 
> I hope that I've taught myself enough over the last few weeks that this isn't a blindingly stupid question?
> 
> I am providing services to multiple customers on FS 1.0.6
> 
> I have a basic setup; (which I have inherited, hence the basic questions);
> 
> AS5350--------<external>----- FS-----<internal>------customer GW(s)  
> (No Registration)							(No Registration)
> 
> All of the GW info is pulled from a PSWL database. Carriers, carrier_id, lcr table etc.
> 
> The bit I am missing is;
> 
> If all of my GW config is in the PSQL database, and that's all accessed by mod_lcr before routing the call, how do I allow for different DTMF types, codecs etc, PER CUSTOMER GW ?
> 
> Am I stuck with actually defining the gateways statically under /etc/freeswitch/conf/sip_profiles/ (or I see I can doit in the dialplan, but that won't achieve what I need)
> 
> I was hoping to be able to pull the DTMF (and other misc info) from the database, but the more I look into it,  the less likely it is that this will work I think. (mod_lcr is already at the routing stage, can't modify DTMF at that point AFAIK)
> 
> Like I said, hope the question isn't so obvious that it's crazy?
> 
> Regards,
> Mike
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From steveayre at gmail.com  Wed Jun  8 11:51:18 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 8 Jun 2011 08:51:18 +0100
Subject: [Freeswitch-users] ACL
In-Reply-To: <4DEEBF6D.7050608@gmail.com>
References: <4DEEBF6D.7050608@gmail.com>
Message-ID: <BANLkTimGGNr1u-rS3F4hyk2oYebHvz-6=A@mail.gmail.com>

   <list name="strict" *default="allow"*>
      <node type="allow" cidr="195.225.XXX.XXX/32"/>
    </list>

You want a default of 'deny'. The default is what to do with any IP not on
the list, what you've set allows every IP.

I would suggest you build your config off the default config by the way,
customising them to suit you and removing what you don't need. The directory
structure is there to help you manage things (what you've posted makes it
look like you might be using a flat file?). There's a lot of parameters
you've missed out setting.

For example:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h
,G722,PCMU,PCMA,GSM"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>

These will have absolutely no effect. They set a global variable that you
can use elsewhere in your configuration, but you're not actually using them.
You should be setting the inbound & outbound codec preferences on the sofia
profile in a param, either using the global variable e.g.
$${outbound_codec_prefs} or by setting them explicitly there.

-Steve


On 8 June 2011 01:16, Cyril Zlachevsky <cyril.zlachevsky at gmail.com> wrote:

> Hi,
> Can't stand what should I do for ACL working.
>
> I'm forwarding a call from a particular static IP to number at voipprovider.
> I refused of using FS default configs because they are complicated and
> redundant for me.
>
> This is my configuration:
> <?xml version="1.0"?>
> <document type="freeswitch/xml">
>
> <X-PRE-PROCESS cmd="set" data="local_address=$${local_ip_v4}"/>
>
> <X-PRE-PROCESS cmd="set" data="sipnet_proxy=voiprovider.com"/>
> <X-PRE-PROCESS cmd="set" data="sipnet_login=echo"/>
> <X-PRE-PROCESS cmd="set" data="sipnet_password=echo"/>
>
> <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/>
>
> <section name="configuration">
>
> <configuration name="switch.conf">
>   <settings>
>     <param name="colorize-console" value="true"/>
>     <param name="max-sessions" value="1000"/>
>     <param name="sessions-per-second" value="30"/>
>     <param name="loglevel" value="debug"/>
>   </settings>
> </configuration>
>
> <configuration name="modules.conf">
>   <modules>
>     <load module="mod_console"/>
>     <load module="mod_logfile"/>
>     <load module="mod_sofia"/>
>     <load module="mod_dialplan_xml"/>
>     <load module="mod_dptools"/>
>     <load module="mod_commands"/>
>     <load module="mod_expr"/>
>     <load module="mod_sndfile"/>
>   </modules>
> </configuration>
>
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h
> ,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>
> <configuration name="console.conf">
>   <mappings>
>     <map name="all"
> value="console,debug,info,notice,warning,err,crit,alert"/>
>   </mappings>
>   <settings>
>     <param name="colorize" value="true"/>
>     <param name="loglevel" value="info"/>
>   </settings>
> </configuration>
>
> <configuration name="logfile.conf">
>   <settings>
>     <param name="rotate-on-hup" value="true"/>
>   </settings>
>   <profiles>
>     <profile name="default">
>       <settings>
>         <param name="logfile" value="/var/log/freeswitch/freeswitch.log"/>
>         <param name="rollover" value="10485760"/>
>       </settings>
>       <mappings>
>         <map name="all" value="debug,info,notice,warning,err,crit,alert"/>
>       </mappings>
>     </profile>
>   </profiles>
> </configuration>
>
> <configuration name="acl.conf" description="Network Lists">
>   <network-lists>
>     <list name="strict" default="allow">
>       <node type="allow" cidr="195.225.XXX.XXX/32"/>
>     </list>
>     <list name="domains" default="deny">
>       <node type="allow" domain="$${domain}"/>
>     </list>
>   </network-lists>
> </configuration>
>
> <configuration name="sofia.conf">
>   <global_settings>
>     <param name="log-level" value="0"/>
>     <param name="debug-presence" value="0"/>
>   </global_settings>
>   <profiles>
>     <profile name="local">
>       <domains>
>         <domain name="all" alias="true" parse="false"/>
>       </domains>
>       <settings>
>         <param name="context" value="local"/>
>         <param name="sip-port" value="5060"/>
>         <param name="dialplan" value="XML"/>
>         <param name="rtp-ip" value="$${local_address}"/>
>         <param name="sip-ip" value="$${local_address}"/>
>       </settings>
>     </profile>
>     <profile name="sipnet">
>       <domains>
>         <domain name="all" alias="false" parse="false"/>
>       </domains>
>       <settings>
>         <param name="context" value="sipnet"/>
>         <param name="sip-port" value="5080"/>
>         <param name="dialplan" value="XML"/>
>         <param name="rtp-ip" value="$${local_address}"/>
>         <param name="sip-ip" value="$${local_address}"/>
>         <!--param name="auth-calls" value="false"/-->
>         <param name="auth-calls" value="true"/>
>         <param name="apply-inbound-acl" value="strict"/>
>       </settings>
>       <gateways>
>         <gateway name="sipnet">
>           <param name="proxy" value="$${sipnet_proxy}"/>
>           <param name="realm" value="asterisk"/>
>           <param name="username" value="$${sipnet_login}"/>
>           <param name="password" value="$${sipnet_password}"/>
>           <param name="local-network-acl" value="strict"/>
>         </gateway>
>       </gateways>
>     </profile>
>   </profiles>
>  </configuration>
> </section>
>
> <section name="dialplan">
>   <context name="local">
>     <extension name="local-accounts">
>       <condition field="destination_number" expression="^(10[1-3])$">
>         <action application="bridge" data="user/$1@$${local_address}"/>
>       </condition>
>     </extension>
>     <extension name="sipnet">
>       <condition field="destination_number" expression="^(\d+)$">
>         <action application="set"
> data="effective_caller_id_number=$${sipnet_login}"/>
>         <action application="bridge"
> data="{sip_invite_domain=$${sipnet_proxy}}sofia/sipnet/$1@
> $${sipnet_proxy}"/>
>       </condition>
>     </extension>
>   </context>
> </section>
>
> <section name="directory">
>   <domain name="$${local_address}">
>     <params>
>       <param name="dial-string"
> value="{presence_id=${dialed_user}@
> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>     </params>
>     <groups>
>       <group name="local">
>         <users>
>           <user id="inboundtest"></user>
>           <params>
>             <param name="debug" value="yes"/>
>             <param name="register" value="true"/>
>             <param name="apply-inbound-acl" value="strict"/>
>             <param name="apply-register-acl" value="strict"/>
>           </params>
>          </users>
>       </group>
>     </groups>
>   </domain>
> </section>
> </document>
>
> When I start FS, I can't see my IP 195.225.XXX.XXX in freeswitch.log - only
> this:
> [NOTICE] switch_core.c:1088 Created ip list rfc1918.auto default (deny)
> [NOTICE] switch_utils.c:248 Adding 10.0.0.0/8 (allow) [] to list
> rfc1918.auto
> [NOTICE] switch_utils.c:248 Adding 172.16.0.0/12 (allow) [] to list
> rfc1918.auto
> [NOTICE] switch_utils.c:248 Adding 192.168.0.0/16 (allow) [] to list
> rfc1918.auto
> [NOTICE] switch_core.c:1096 Created ip list wan.auto default (allow)
> [NOTICE] switch_utils.c:248 Adding 10.0.0.0/8 (deny) [] to list wan.auto
> [NOTICE] switch_utils.c:248 Adding 172.16.0.0/12 (deny) [] to list
> wan.auto
> [NOTICE] switch_utils.c:248 Adding 192.168.0.0/16 (deny) [] to list
> wan.auto
> [NOTICE] switch_core.c:1104 Created ip list nat.auto default (deny)
> [NOTICE] switch_core.c:1106 Adding 88.198.XXX.XXX/255.255.255.255 (deny)
> to list nat.auto
> [NOTICE] switch_utils.c:248 Adding 10.0.0.0/8 (allow) [] to list nat.auto
> [NOTICE] switch_utils.c:248 Adding 172.16.0.0/12 (allow) [] to list
> nat.auto
> [NOTICE] switch_utils.c:248 Adding 192.168.0.0/16 (allow) [] to list
> nat.auto
> [NOTICE] switch_core.c:1115 Created ip list loopback.auto default (deny)
> [NOTICE] switch_utils.c:248 Adding 127.0.0.0/8 (allow) [] to list
> loopback.auto
> [NOTICE] switch_core.c:1121 Created ip list localnet.auto default (deny)
> [NOTICE] switch_core.c:1124 Adding 88.198.XXX.XXX/255.255.255.255 (allow)
> to list localnet.auto
>
> With my current configuration FS allow to register from any IP.
> Where is my error?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Wed Jun  8 11:59:35 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 8 Jun 2011 08:59:35 +0100
Subject: [Freeswitch-users] Conceptual/design question - per
 gateway/DTMF & PSQL
In-Reply-To: <CA152D95.2FFE8%mike.taylor@staff.snap.net.nz>
References: <CA152D11.2FFDF%mike.taylor@staff.snap.net.nz>
	<CA152D95.2FFE8%mike.taylor@staff.snap.net.nz>
Message-ID: <BANLkTikcWbjCHnpefni6cL7ttY5p3On_sw@mail.gmail.com>

The dtmf_type channel variable can be used to override the profile's dtmf
setting. Valid settings are rf2833, info and none. Inband would be more
complex, as you'd need to call the start_dtmf_generate application, not sure
how you could easily integrate that with mod_lcr. Possibly using the
execute_on_x family of channel variables. mod_lcr would need to use custom
sql to get the additional dtmf variable.

As David pointed out, mod_lcr already lets you set the codec list for a
gateway. late-negotiation isn't *required*, but could be useful. Without it
the aleg will pick a codec, then you'll dial out with the codec list to the
gateway and transcode if needed. With it, the aleg doesn't pick the codec
until the bleg has picked one - which may make transcoding less likely.

You could also setup the gateways as sofia gateways, set the codec/dtmf
variables on the gateway and have mod_lcr return the sofia/gateway/gwname/
prefix.

A lot of the behaviour in terms of mod_lcr, execute_on_x etc has changed
since 1.0.6 so you might find you have to upgrade to get some of the newer
features.

-Steve


On 8 June 2011 02:46, Mike Taylor <mtaylor at employees.org> wrote:

> Hi,
>
> I hope that I've taught myself enough over the last few weeks that this
> isn't a blindingly stupid question?
>
> I am providing services to multiple customers on FS 1.0.6
>
> I have a basic setup; (which I have inherited, hence the basic questions);
>
> AS5350--------<external>----- FS-----<internal>------customer GW(s)
> (No Registration) (No Registration)
>
> All of the GW info is pulled from a PSWL database. Carriers, carrier_id,
> lcr table etc.
>
> The bit I am missing is;
>
> If all of my GW config is in the PSQL database, and that's all accessed by
> mod_lcr before routing the call, how do I allow for different DTMF types,
> codecs etc, PER CUSTOMER GW ?
>
> Am I stuck with actually defining the gateways statically under
> /etc/freeswitch/conf/sip_profiles/ (or I see I can doit in the dialplan, but
> that won't achieve what I need)
>
> I was hoping to be able to pull the DTMF (and other misc info) from the
> database, but the more I look into it,  the less likely it is that this will
> work I think. (mod_lcr is already at the routing stage, can't modify DTMF at
> that point AFAIK)
>
> Like I said, hope the question isn't so obvious that it's crazy?
>
> Regards,
> Mike
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Wed Jun  8 13:01:37 2011
From: u2nsam at gmail.com (Sam)
Date: Wed, 8 Jun 2011 14:31:37 +0530
Subject: [Freeswitch-users] domain
In-Reply-To: <4DE9D5E7.7030800@tagnet.ru>
References: <BANLkTimO7+Sg=8cJjdfRr-1wEKgbbZDrcA@mail.gmail.com>
	<4DE89971.9010205@tagnet.ru>
	<BANLkTimOZ-yxbNXFiOr519FoR=9euOZLcg@mail.gmail.com>
	<4DE8A680.1080102@tagnet.ru>
	<BANLkTinkghJTnEw16ED7PfnbmUBAPk=9dA@mail.gmail.com>
	<4DE9D5E7.7030800@tagnet.ru>
Message-ID: <BANLkTikAGnnYmrW_GuYP+A+1yEQEunW9dw@mail.gmail.com>

Thankyou.

Regds
Sam

On Sat, Jun 4, 2011 at 12:21 PM, Boris Kovalenko <boris at tagnet.ru> wrote:

>  Hello!
>
>     No, You can't use it this way. There are 2 possibilites:
> 1. You have one context (dialplan) and use something like
>
> <extension name="ext_01">
>     <condition field="${domain_name}" expression="^xyz.com$"/>
>     <condition filed="destination_number" expression="^(1001)$">
>
>         <action application="voicemail" data="default $${domain name}
> ${sip_to_host}"/>
>     </condition>
> </extension>
>
> 2. I prefer the way to have separate contexts (dialplans) for each domain.
> So in the main context You do something like:
> <action application="tranfser" data="${destination_number} XML
> context_${domain_name}"/>
> And after that You know that is context_${domain_name} (context_xyz.com as
> in your example) You have o
>
> Hello,
>
> After doing this ,
>
> <action application="set" data="domain
> name=${regex(${sip_from_uri}|^.*@(.*)$|%1)}" inline="true"/>
>
> i get domain name = xyz.com
>
> Now how can i use the fundas of identified domain and using below to
> execute .
>
> <domain name="xyz.com">
> <condition field="destination_number" expression="^(1001)$">
>  <action application="voicemail" data="default $${domain name}
> ${sip_to_host}"/>
> </domain>
>
>
>
> Is that possible.
>
>
> Regards
> Sam
>
>
>  On Fri, Jun 3, 2011 at 2:46 PM, Boris Kovalenko <boris at tagnet.ru> wrote:
>
>>  Hello!
>>
>>     There is a variable named ${sip_from_user}. You may do something like
>>
>> <action application="set" data=
>> "from_domain=${regex(${sip_from_user}|^.*@(.*)$|%1)}"<from_domain=$%7Bregex%28$%7Bsip_from_user%7D%7C%5E.*@%28.*%29$%7C%251%29%7D>inline="true"/>
>> <action application="transfer" data="${from_domain} XML domain_context"/>
>>
>>
>> The call would be coming from outside to FS and not within , so the call
>> would be routed from public to different domains ,
>>
>> and after that following the doc
>> http://wiki.freeswitch.org/wiki/Multi-tenant
>>
>> Here how can we recognized from the header on which domain is it ?  like
>> if we get an from header as 1001 at xyz.com
>> how can we recognized the domain and route the call accordingly ... when
>> an outside system sends a call via acl .
>>
>> regards
>> Sam
>>
>> On Fri, Jun 3, 2011 at 1:51 PM, Boris Kovalenko <boris at tagnet.ru> wrote:
>>
>>>  Hello!
>>>
>>>     What do You mean? You may (for example) set the domain variable for a
>>> user and test it inside condition.
>>>
>>>  Hello,
>>>
>>> I have a user registered by domain (1001 at xyz.com) and when the invite
>>> comes to my FS server it should recognize the domain and route
>>> the calls accordingly to the group <domain name="xyz.com">  ...
>>> </domain> .
>>> The call would be entering via the acl to public.xml. Here i want to
>>> route the call according to domain names, how will i do that.
>>>
>>> Any suggestions...
>>>
>>>
>>> Regards
>>> Sam
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>
>>>
>>>
>>> --
>>> Regards,
>>> Boris
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> --
>> Regards,
>> Boris
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> Regards,
> Boris
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Wed Jun  8 13:21:50 2011
From: u2nsam at gmail.com (Sam)
Date: Wed, 8 Jun 2011 14:51:50 +0530
Subject: [Freeswitch-users] domain
In-Reply-To: <4DE9D5E7.7030800@tagnet.ru>
References: <BANLkTimO7+Sg=8cJjdfRr-1wEKgbbZDrcA@mail.gmail.com>
	<4DE89971.9010205@tagnet.ru>
	<BANLkTimOZ-yxbNXFiOr519FoR=9euOZLcg@mail.gmail.com>
	<4DE8A680.1080102@tagnet.ru>
	<BANLkTinkghJTnEw16ED7PfnbmUBAPk=9dA@mail.gmail.com>
	<4DE9D5E7.7030800@tagnet.ru>
Message-ID: <BANLkTimy3i_b=AzoWE6WXz5mA=aQLCYH8A@mail.gmail.com>

Had it been FS identifying the call to be of some domain and using the
dial-plan with <domain name="xyz.com">  ...   </domain> ,
that would had been great .

Regds
Sam



On Sat, Jun 4, 2011 at 12:21 PM, Boris Kovalenko <boris at tagnet.ru> wrote:

>  Hello!
>
>     No, You can't use it this way. There are 2 possibilites:
> 1. You have one context (dialplan) and use something like
>
> <extension name="ext_01">
>     <condition field="${domain_name}" expression="^xyz.com$"/>
>     <condition filed="destination_number" expression="^(1001)$">
>
>         <action application="voicemail" data="default $${domain name}
> ${sip_to_host}"/>
>     </condition>
> </extension>
>
> 2. I prefer the way to have separate contexts (dialplans) for each domain.
> So in the main context You do something like:
> <action application="tranfser" data="${destination_number} XML
> context_${domain_name}"/>
> And after that You know that is context_${domain_name} (context_xyz.com as
> in your example) You have o
>
> Hello,
>
> After doing this ,
>
> <action application="set" data="domain
> name=${regex(${sip_from_uri}|^.*@(.*)$|%1)}" inline="true"/>
>
> i get domain name = xyz.com
>
> Now how can i use the fundas of identified domain and using below to
> execute .
>
> <domain name="xyz.com">
> <condition field="destination_number" expression="^(1001)$">
>  <action application="voicemail" data="default $${domain name}
> ${sip_to_host}"/>
> </domain>
>
>
>
> Is that possible.
>
>
> Regards
> Sam
>
>
>  On Fri, Jun 3, 2011 at 2:46 PM, Boris Kovalenko <boris at tagnet.ru> wrote:
>
>>  Hello!
>>
>>     There is a variable named ${sip_from_user}. You may do something like
>>
>> <action application="set" data=
>> "from_domain=${regex(${sip_from_user}|^.*@(.*)$|%1)}"<from_domain=$%7Bregex%28$%7Bsip_from_user%7D%7C%5E.*@%28.*%29$%7C%251%29%7D>inline="true"/>
>> <action application="transfer" data="${from_domain} XML domain_context"/>
>>
>>
>> The call would be coming from outside to FS and not within , so the call
>> would be routed from public to different domains ,
>>
>> and after that following the doc
>> http://wiki.freeswitch.org/wiki/Multi-tenant
>>
>> Here how can we recognized from the header on which domain is it ?  like
>> if we get an from header as 1001 at xyz.com
>> how can we recognized the domain and route the call accordingly ... when
>> an outside system sends a call via acl .
>>
>> regards
>> Sam
>>
>> On Fri, Jun 3, 2011 at 1:51 PM, Boris Kovalenko <boris at tagnet.ru> wrote:
>>
>>>  Hello!
>>>
>>>     What do You mean? You may (for example) set the domain variable for a
>>> user and test it inside condition.
>>>
>>>  Hello,
>>>
>>> I have a user registered by domain (1001 at xyz.com) and when the invite
>>> comes to my FS server it should recognize the domain and route
>>> the calls accordingly to the group <domain name="xyz.com">  ...
>>> </domain> .
>>> The call would be entering via the acl to public.xml. Here i want to
>>> route the call according to domain names, how will i do that.
>>>
>>> Any suggestions...
>>>
>>>
>>> Regards
>>> Sam
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>
>>>
>>>
>>> --
>>> Regards,
>>> Boris
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> --
>> Regards,
>> Boris
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> Regards,
> Boris
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From boris at tagnet.ru  Wed Jun  8 13:42:55 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Wed, 08 Jun 2011 15:42:55 +0600
Subject: [Freeswitch-users] domain
In-Reply-To: <BANLkTimy3i_b=AzoWE6WXz5mA=aQLCYH8A@mail.gmail.com>
References: <BANLkTimO7+Sg=8cJjdfRr-1wEKgbbZDrcA@mail.gmail.com>	<4DE89971.9010205@tagnet.ru>	<BANLkTimOZ-yxbNXFiOr519FoR=9euOZLcg@mail.gmail.com>	<4DE8A680.1080102@tagnet.ru>	<BANLkTinkghJTnEw16ED7PfnbmUBAPk=9dA@mail.gmail.com>	<4DE9D5E7.7030800@tagnet.ru>
	<BANLkTimy3i_b=AzoWE6WXz5mA=aQLCYH8A@mail.gmail.com>
Message-ID: <4DEF441F.3070904@tagnet.ru>

Hello!

     Hmmm... I think this is useless feature. You already have an 
user_context and separate sip profiles.

> Had it been FS identifying the call to be of some domain and using the 
> dial-plan with <domain name="xyz.com <http://xyz.com/>">  ... </domain> ,
> that would had been great .
>
> Regds
> Sam
>
>
>
> On Sat, Jun 4, 2011 at 12:21 PM, Boris Kovalenko <boris at tagnet.ru 
> <mailto:boris at tagnet.ru>> wrote:
>
>     Hello!
>
>         No, You can't use it this way. There are 2 possibilites:
>     1. You have one context (dialplan) and use something like
>
>     <extension name="ext_01">
>     <condition field="${domain_name}" expression="^xyz.com
>     <http://xyz.com>$"/>
>     <condition filed="destination_number" expression="^(1001)$">
>
>     <action application="voicemail" data="default $${domain name}
>     ${sip_to_host}"/>
>     </condition>
>     </extension>
>
>     2. I prefer the way to have separate contexts (dialplans) for each
>     domain. So in the main context You do something like:
>     <action application="tranfser" data="${destination_number} XML
>     context_${domain_name}"/>
>     And after that You know that is context_${domain_name}
>     (context_xyz.com <http://context_xyz.com> as in your example) You
>     have o
>
>>     Hello,
>>
>>     After doing this ,
>>
>>     <action application="set" data="domain
>>     name=${regex(${sip_from_uri}|^.*@(.*)$|%1)}" inline="true"/>
>>
>>     i get domain name = xyz.com <http://xyz.com>
>>
>>     Now how can i use the fundas of identified domain and using below
>>     to execute .
>>
>>     <domain name="xyz.com <http://xyz.com>">
>>     <condition field="destination_number" expression="^(1001)$">
>>     <action application="voicemail" data="default $${domain name}
>>     ${sip_to_host}"/>
>>     </domain>
>>
>>
>>
>>     Is that possible.
>>
>>
>>     Regards
>>     Sam
>>
>>
>>     On Fri, Jun 3, 2011 at 2:46 PM, Boris Kovalenko <boris at tagnet.ru
>>     <mailto:boris at tagnet.ru>> wrote:
>>
>>         Hello!
>>
>>             There is a variable named ${sip_from_user}. You may do
>>         something like
>>
>>         <action application="set"
>>         data="from_domain=${regex(${sip_from_user}|^.*@(.*)$|%1)}"
>>         <mailto:from_domain=$%7Bregex%28$%7Bsip_from_user%7D%7C%5E.*@%28.*%29$%7C%251%29%7D>
>>         inline="true"/>
>>         <action application="transfer" data="${from_domain} XML
>>         domain_context"/>
>>
>>
>>>         The call would be coming from outside to FS and not within ,
>>>         so the call would be routed from public to different domains ,
>>>
>>>         and after that following the doc
>>>         http://wiki.freeswitch.org/wiki/Multi-tenant
>>>
>>>         Here how can we recognized from the header on which domain
>>>         is it ?  like if we get an from header as 1001 at xyz.com
>>>         <mailto:1001 at xyz.com>
>>>         how can we recognized the domain and route the call
>>>         accordingly ... when an outside system sends a call via acl .
>>>
>>>         regards
>>>         Sam
>>>
>>>         On Fri, Jun 3, 2011 at 1:51 PM, Boris Kovalenko
>>>         <boris at tagnet.ru <mailto:boris at tagnet.ru>> wrote:
>>>
>>>             Hello!
>>>
>>>                 What do You mean? You may (for example) set the
>>>             domain variable for a user and test it inside condition.
>>>
>>>>             Hello,
>>>>
>>>>             I have a user registered by domain (1001 at xyz.com
>>>>             <mailto:1001 at xyz.com>) and when the invite comes to my
>>>>             FS server it should recognize the domain and route
>>>>             the calls accordingly to the group <domain
>>>>             name="xyz.com <http://xyz.com>">  ... </domain> .
>>>>             The call would be entering via the acl to public.xml.
>>>>             Here i want to route the call according to domain
>>>>             names, how will i do that.
>>>>
>>>>             Any suggestions...
>>>>
>>>>
>>>>             Regards
>>>>             Sam
>>>>
>>>>
>>>>
>>>>             _______________________________________________
>>>>             FreeSWITCH-users mailing list
>>>>             FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>             http://www.freeswitch.org
>>>
>>>
>>>             -- 
>>>             Regards,
>>>             Boris
>>>
>>>
>>>
>>>             _______________________________________________
>>>             FreeSWITCH-users mailing list
>>>             FreeSWITCH-users at lists.freeswitch.org
>>>             <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>             http://www.freeswitch.org
>>>
>>>
>>>
>>>         _______________________________________________
>>>         FreeSWITCH-users mailing list
>>>         FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>         http://www.freeswitch.org
>>
>>
>>         -- 
>>         Regards,
>>         Boris
>>
>>
>>
>>         _______________________________________________
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>     -- 
>     Regards,
>     Boris
>
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Regards,
Boris


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From sharad at coraltele.com  Wed Jun  8 16:06:47 2011
From: sharad at coraltele.com (sharad)
Date: Wed, 8 Jun 2011 17:36:47 +0530
Subject: [Freeswitch-users] Voicemail Message - How t know how many are read
	& how many are unread in Inbox
References: <mailman.10243.1307524956.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <AA1BE0FA88F3421784DBD1D91A84E515@sharad>

Dear All,

Just wondering whether there is any way to know how many voice messages are 
read or unread in New & SAVED box.

A help will be highly appriciated.

Regards
Sharad 



From valery.kalinin at gmail.com  Wed Jun  8 16:18:29 2011
From: valery.kalinin at gmail.com (Valery Kalinin)
Date: Wed, 8 Jun 2011 18:18:29 +0600
Subject: [Freeswitch-users] FreeTDM does not work
Message-ID: <BANLkTimQnSg5E9aKYeVvfcxxkeiQGvRRHw@mail.gmail.com>

Hi all!

FreeTDM does not work with Digium equipment.

Equipment: Digium TE121
OS: CentOS 5.6
DAHDI: 2.4.1.2
# ./dahdi_scan
[1]
active=yes
alarms=OK
description=Wildcard TE121 Card 0
name=WCT1/0
manufacturer=Digium
devicetype=Wildcard TE121 (VPMADT032)
location=PCI Bus 04 Slot 09
basechan=1
totchans=31
irq=5
type=digital-E1
syncsrc=1
lbo=0 db (CSU)/0-133 feet (DSX-1)
coding_opts=AMI,HDB3
framing_opts=CCS,CRC4
coding=HDB3
framing=CCS

LIBPRI: 1.4.11.5

zt.conf
[defaults]
codec_ms => 20
wink_ms => 150
flash_ms => 750
echo_cancel_level => 64
rxgain => 0.0
txgain => 0.0

freetdm.conf
[span zt pri]
name => pri
trunk_type => E1
group => e1group
b-channel => 1-15
b-channel => 17-31
d-channel => 16

part of freetdm.conf.xml:
    <libpri_spans>
        <span id="1" name="pri">
            <param name="node" value="network"/>
            <param name="switch" value="4ess"/>
            <param name="l1" value="alaw"/>
            <param name="dp" value="unknown"/>
            <param name="debug" value="q931_all"/>
            <param name="dialplan" value="XML"/>
            <param name="context" value="default"/>
        </span>
    </libpri_spans>

and log result (important parts):
start freetdm:
2011-06-09 02:07:00.591482 [DEBUG] ftdm_config.c:52 New mod directory:
/usr/local/freeswitch/mod
2011-06-09 02:07:00.591492 [DEBUG] ftdm_config.c:58 New config directory:
/usr/local/freeswitch/conf
2011-06-09 02:07:00.591539 [DEBUG] ftdm_sched.c:154 Initializing scheduling
API
2011-06-09 02:07:00.591545 [DEBUG] ftdm_sched.c:251 Created schedule
freetdm-master
2011-06-09 02:07:00.591551 [NOTICE] ftdm_sched.c:178 Launching main schedule
thread
2011-06-09 02:07:00.591581 [DEBUG] ftdm_sched.c:187 Running schedule
freetdm-master in the main schedule thread
2011-06-09 02:07:00.591592 [DEBUG] ftdm_config.c:80 Configuration file is
/usr/local/freeswitch/conf/modules.conf.
2011-06-09 02:07:00.591639 [NOTICE] ftdm_io.c:5656 Modules configured: 1
2011-06-09 02:07:00.591645 [DEBUG] ftdm_config.c:80 Configuration file is
/usr/local/freeswitch/conf/freetdm.conf.
2011-06-09 02:07:00.591659 [DEBUG] ftdm_io.c:4595 Reading FreeTDM
configuration file
2011-06-09 02:07:00.591679 [DEBUG] ftdm_io.c:4611 found config for span
2011-06-09 02:07:00.591783 [NOTICE] ftmod_zt.c:1300 Using DAHDI control
device
2011-06-09 02:07:00.591794 [INFO] ftdm_io.c:4898 Loading IO from
/usr/local/freeswitch/mod/ftmod_zt.so [zt]
2011-06-09 02:07:00.591814 [DEBUG] ftdm_config.c:80 Configuration file is
/usr/local/freeswitch/conf/zt.conf.
2011-06-09 02:07:00.591840 [INFO] ftmod_zt.c:577 Setting rxgain val to
0.000000
2011-06-09 02:07:00.591846 [INFO] ftmod_zt.c:585 Setting txgain val to
0.000000
2011-06-09 02:07:00.591860 [INFO] ftdm_io.c:790 Auto-loaded I/O module 'zt'
2011-06-09 02:07:00.591885 [DEBUG] ftdm_io.c:4625 created span 1 (pri) of
type zt
2011-06-09 02:07:00.591890 [DEBUG] ftdm_io.c:4641 span 1 [name]=[pri]
2011-06-09 02:07:00.591896 [DEBUG] ftdm_io.c:4641 span 1 [trunk_type]=[E1]
2011-06-09 02:07:00.591902 [DEBUG] ftdm_io.c:4646 setting trunk type to 'E1'
2011-06-09 02:07:00.591907 [DEBUG] ftdm_io.c:4641 span 1 [group]=[e1group]
2011-06-09 02:07:00.591912 [DEBUG] ftdm_io.c:4641 span 1 [b-channel]=[1-15]
2011-06-09 02:07:00.591940 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 1 as FreeTDM device 1:1 fd:25
2011-06-09 02:07:00.591963 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 2 as FreeTDM device 1:2 fd:26
2011-06-09 02:07:00.591984 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 3 as FreeTDM device 1:3 fd:27
2011-06-09 02:07:00.592006 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 4 as FreeTDM device 1:4 fd:28
2011-06-09 02:07:00.592029 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 5 as FreeTDM device 1:5 fd:29
2011-06-09 02:07:00.592049 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 6 as FreeTDM device 1:6 fd:30
2011-06-09 02:07:00.592069 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 7 as FreeTDM device 1:7 fd:31
2011-06-09 02:07:00.592090 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 8 as FreeTDM device 1:8 fd:32
2011-06-09 02:07:00.592110 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 9 as FreeTDM device 1:9 fd:33
2011-06-09 02:07:00.592131 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 10 as FreeTDM device 1:10 fd:34
2011-06-09 02:07:00.592151 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 11 as FreeTDM device 1:11 fd:35
2011-06-09 02:07:00.592171 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 12 as FreeTDM device 1:12 fd:36
2011-06-09 02:07:00.592192 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 13 as FreeTDM device 1:13 fd:37
2011-06-09 02:07:00.592212 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 14 as FreeTDM device 1:14 fd:38
2011-06-09 02:07:00.592231 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 15 as FreeTDM device 1:15 fd:39
2011-06-09 02:07:00.592239 [DEBUG] ftdm_io.c:5200 Creating new group:e1group
2011-06-09 02:07:00.592275 [DEBUG] ftdm_io.c:4641 span 1 [b-channel]=[17-31]
2011-06-09 02:07:00.592299 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 17 as FreeTDM device 1:16 fd:40
2011-06-09 02:07:00.592323 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 18 as FreeTDM device 1:17 fd:41
2011-06-09 02:07:00.592348 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 19 as FreeTDM device 1:18 fd:42
2011-06-09 02:07:00.592369 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 20 as FreeTDM device 1:19 fd:43
2011-06-09 02:07:00.592390 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 21 as FreeTDM device 1:20 fd:44
2011-06-09 02:07:00.592422 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 22 as FreeTDM device 1:21 fd:45
2011-06-09 02:07:00.592442 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 23 as FreeTDM device 1:22 fd:46
2011-06-09 02:07:00.592462 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 24 as FreeTDM device 1:23 fd:47
2011-06-09 02:07:00.592486 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 25 as FreeTDM device 1:24 fd:48
2011-06-09 02:07:00.592510 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 26 as FreeTDM device 1:25 fd:49
2011-06-09 02:07:00.592531 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 27 as FreeTDM device 1:26 fd:50
2011-06-09 02:07:00.592558 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 28 as FreeTDM device 1:27 fd:51
2011-06-09 02:07:00.592584 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 29 as FreeTDM device 1:28 fd:52
2011-06-09 02:07:00.592609 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 30 as FreeTDM device 1:29 fd:53
2011-06-09 02:07:00.592628 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 31 as FreeTDM device 1:30 fd:54
2011-06-09 02:07:00.592654 [DEBUG] ftdm_io.c:4641 span 1 [d-channel]=[16]
2011-06-09 02:07:00.592692 [INFO] ftmod_zt.c:396 configuring device
/dev/dahdi/channel channel 16 as FreeTDM device 1:31 fd:55
2011-06-09 02:07:00.592712 [INFO] ftdm_io.c:4823 Configured 31 channel(s)
2011-06-09 02:07:00.593269 [INFO] ftdm_io.c:4898 Loading IO from
/usr/local/freeswitch/mod/ftmod_libpri.so [libpri]
2011-06-09 02:07:00.593283 [DEBUG] ftdm_io.c:4839 Module libpri does not
support configuration.
2011-06-09 02:07:00.593287 [INFO] ftdm_io.c:4910 Loading SIG from
/usr/local/freeswitch/mod/ftmod_libpri.so
2011-06-09 02:07:00.593292 [INFO] ftdm_io.c:5133 auto-loaded 'libpri'
2011-06-09 02:07:00.593302 [NOTICE] ftmod_libpri.c:1981 Setting default
Layer 1 to ALAW since this is an E1/BRI/BRI PTMP trunk
2011-06-09 02:07:00.593566 [DEBUG] ftmod_libpri.c:1629 opening D-Channel #0
1:31
2011-06-09 02:07:00.596450 [NOTICE] switch_loadable_module.c:145 Adding
Endpoint 'freetdm'
2011-06-09 02:07:00.596481 [NOTICE] switch_loadable_module.c:252 Adding
Application 'disable_ec'
2011-06-09 02:07:00.596510 [NOTICE] switch_loadable_module.c:252 Adding
Application 'disable_dtmf'
2011-06-09 02:07:00.596530 [NOTICE] switch_loadable_module.c:252 Adding
Application 'enable_dtmf'
2011-06-09 02:07:00.596552 [NOTICE] switch_loadable_module.c:274 Adding API
Function 'ftdm'
2011-06-09 02:07:00.618807 [DEBUG] ftmod_libpri.c:1586 -- Caught Event span
1 4 (CONFIG_ERR)
2011-06-09 02:07:00.622776 [NOTICE] switch_loadable_module.c:274 Adding API
Function 'acl'
...later:
2011-06-09 02:07:01.693551 [DEBUG] ftmod_libpri.c:1586 -- Caught Event span
1 4 (CONFIG_ERR)
2011-06-09 02:07:02.773590 [DEBUG] ftmod_libpri.c:1586 -- Caught Event span
1 4 (CONFIG_ERR)
2011-06-09 02:07:03.833629 [DEBUG] ftmod_libpri.c:1586 -- Caught Event span
1 4 (CONFIG_ERR)
2011-06-09 02:07:04.893667 [DEBUG] ftmod_libpri.c:146 TEI=0 DL event:
Q931_DL_EVENT_DL_RELEASE_IND(3)
2011-06-09 02:07:05.962709 [DEBUG] ftmod_libpri.c:1586 -- Caught Event span
1 4 (CONFIG_ERR)
2011-06-09 02:07:07.982780 [DEBUG] ftmod_libpri.c:1586 -- Caught Event span
1 4 (CONFIG_ERR)
2011-06-09 02:07:10.002854 [DEBUG] ftmod_libpri.c:1586 -- Caught Event span
1 4 (CONFIG_ERR)
...
making a call:
...
Dialplan: sofia/internal/1000 at 192.168.200.11 Action
bridge(freetdm/e1group/1/12345)
Dialplan: sofia/internal/1000 at 192.168.200.11 Action answer()
2011-06-09 02:10:27.343089 [DEBUG] mod_dptools.c:1063 sofia/internal/
1000 at 192.168.200.11 SET [RFC2822_DATE]=[Thu, 09 Jun 2011 02:10:27 +0600]
EXECUTE sofia/internal/1000 at 192.168.200.11 bridge(freetdm/e1group/1/12345)
2011-06-09 02:10:27.343089 [DEBUG] switch_ivr_originate.c:1869 Parsing
global variables
2011-06-09 02:10:27.343089 [NOTICE] mod_freetdm.c:1551 Close Channel N/A
[CS_NEW]
2011-06-09 02:10:27.343089 [DEBUG] switch_core_state_machine.c:457 ()
Running State Change CS_DESTROY
2011-06-09 02:10:27.343089 [DEBUG] switch_core_state_machine.c:467 (N/A)
State DESTROY
2011-06-09 02:10:27.343089 [DEBUG] switch_core_state_machine.c:467 (N/A)
State DESTROY going to sleep
2011-06-09 02:10:27.343089 [ERR] switch_ivr_originate.c:2443 Cannot create
outgoing channel of type [freetdm] cause: [NORMAL_CIRCUIT_CONGESTION]
2011-06-09 02:10:27.343089 [DEBUG] switch_ivr_originate.c:3296 Originate
Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]

Why?
Which means error: Caught Event span 1 4 (CONFIG_ERR)
Which config???
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From benkokakao at gmail.com  Wed Jun  8 18:34:31 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Wed, 8 Jun 2011 16:34:31 +0200
Subject: [Freeswitch-users] FreeTDM - destination_number empty on inbound
	call
Message-ID: <BANLkTimCRbEutxEf2vqt6OrbyVwn0XhHdw@mail.gmail.com>

Hi!

My FS is connected to a plain BRI-line via a HFC-S PCI-card. On
inbound calls i don't receive a dialed number. Please see the debug at
http://pastebin.freeswitch.org/16456 for details.
Do i have to fetch the DID from a variable before i can handle the
call or is there some other problem, which i assume?

Outbound calls work just fine btw.

Best regards,
Christian


From mint.office.nick at gmail.com  Wed Jun  8 04:04:25 2011
From: mint.office.nick at gmail.com (Mint.Nick)
Date: Tue, 7 Jun 2011 17:04:25 -0700 (PDT)
Subject: [Freeswitch-users]  LuaSQL, Curl, mod_skypopen,
 mod_dingaling....Can't get 'em to work!
Message-ID: <31778696.post@talk.nabble.com>


Hi all,
I've noticed there isn't much traffic on this forum anymore. I'm having
issues with the things listed in the subj. Is there anyone out there who
might be willing to help a guy figure one or more of these out?

Thanks
-Nick
-- 
View this message in context: http://old.nabble.com/LuaSQL%2C-Curl%2C-mod_skypopen%2C-mod_dingaling....Can%27t-get-%27em-to-work%21-tp31778696p31778696.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From alessandro865 at gmail.com  Wed Jun  8 13:52:42 2011
From: alessandro865 at gmail.com (Alessandro Luppi)
Date: Wed, 8 Jun 2011 11:52:42 +0200
Subject: [Freeswitch-users] FS response
Message-ID: <BANLkTi=y40L2O4rMcfEonPWte1GjgjnWBQ@mail.gmail.com>

Hi,

I have the following scenario:

If the softphone2 respond with 486 freeswitch is working in this way:

softphone1                     FS                  softphone2

INVITE ---------->
                              INVITE --------------->
                              <------------------- 486
<-------------- 200
                              ACK ------------------>
ACK ------------>

<-------------- BYE


i want this behaviour:

softphone1                     FS                  softphone2

INVITE ---------->
                              INVITE --------------->
                              <------------------- 486
<-------------- 486
                              ACK ------------------>
ACK ------------>

Any suggestion?

Regards

Alessandro
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From gmaruzz at gmail.com  Wed Jun  8 19:26:52 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 8 Jun 2011 17:26:52 +0200
Subject: [Freeswitch-users] LuaSQL, Curl, mod_skypopen,
 mod_dingaling....Can't get 'em to work!
In-Reply-To: <31778696.post@talk.nabble.com>
References: <31778696.post@talk.nabble.com>
Message-ID: <BANLkTi=xk8esC+xzo7=5_t5H=BqcXe5g6Q@mail.gmail.com>

First read all the related docs in the wiki: http://wiki.freeswitch.org
Then, search the mailing list archive: lists.freeswitch.org

After that, if you have not found the answers to your doubts, please
write here what is your specific problem, with all the details that
can give us the precise idea of why you are in a situation that is not
documented.

-giovanni

On Wed, Jun 8, 2011 at 2:04 AM, Mint.Nick <mint.office.nick at gmail.com> wrote:
>
> Hi all,
> I've noticed there isn't much traffic on this forum anymore. I'm having
> issues with the things listed in the subj. Is there anyone out there who
> might be willing to help a guy figure one or more of these out?
>
> Thanks
> -Nick
> --
> View this message in context: http://old.nabble.com/LuaSQL%2C-Curl%2C-mod_skypopen%2C-mod_dingaling....Can%27t-get-%27em-to-work%21-tp31778696p31778696.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From yungwei at resolvity.com  Wed Jun  8 19:30:20 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Wed, 8 Jun 2011 11:30:20 -0400
Subject: [Freeswitch-users] An extension with a javascript action can handle
 multiple concurrent calls?
Message-ID: <33095823FD21DF429B481B5163264B7950AC5864D7@VMBX102.ihostexchange.net>

Hi,

I'm wondering if the following extension can handle multiple concurrent calls. Thanks.

  <extension name="test">
        <condition field="destination_number" expression="^9912$">
                <action application="answer"/>
                <action application="javascript" data="test.js"/>
        </condition>
  </extension>


From msc at freeswitch.org  Wed Jun  8 19:32:32 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 8 Jun 2011 08:32:32 -0700
Subject: [Freeswitch-users] An extension with a javascript action can
 handle multiple concurrent calls?
In-Reply-To: <33095823FD21DF429B481B5163264B7950AC5864D7@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950AC5864D7@VMBX102.ihostexchange.net>
Message-ID: <BANLkTi=_OxE7kT7gdSEKK2TCjWQG74CSdw@mail.gmail.com>

Yes, you'll have one js instance for each call. JS does not scale very well,
but if you want the ability to scale a lot then you should use Lua.

-MC

On Wed, Jun 8, 2011 at 8:30 AM, Yungwei Chen <yungwei at resolvity.com> wrote:

> Hi,
>
> I'm wondering if the following extension can handle multiple concurrent
> calls. Thanks.
>
>  <extension name="test">
>        <condition field="destination_number" expression="^9912$">
>                <action application="answer"/>
>                <action application="javascript" data="test.js"/>
>        </condition>
>  </extension>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sos at sokhapkin.dyndns.org  Wed Jun  8 19:40:45 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 8 Jun 2011 11:40:45 -0400
Subject: [Freeswitch-users] An extension with a javascript action can
	handle multiple concurrent calls?
In-Reply-To: <BANLkTi=_OxE7kT7gdSEKK2TCjWQG74CSdw@mail.gmail.com>
References: <33095823FD21DF429B481B5163264B7950AC5864D7@VMBX102.ihostexchange.net>
	<BANLkTi=_OxE7kT7gdSEKK2TCjWQG74CSdw@mail.gmail.com>
Message-ID: <201106081140.45399.sos@sokhapkin.dyndns.org>

To me lua frequently crashes when multiple channels call lua interpreter at 
the same time. Even with trivial lua script which calls single select 
statement using dbh.

On Wednesday 08 June 2011, Michael Collins wrote:
> Yes, you'll have one js instance for each call. JS does not scale very
> well, but if you want the ability to scale a lot then you should use Lua.
> 
> -MC
> 
> On Wed, Jun 8, 2011 at 8:30 AM, Yungwei Chen <yungwei at resolvity.com> wrote:
> > Hi,
> > 
> > I'm wondering if the following extension can handle multiple concurrent
> > calls. Thanks.
> > 
> >  <extension name="test">
> >  
> >        <condition field="destination_number" expression="^9912$">
> >        
> >                <action application="answer"/>
> >                <action application="javascript" data="test.js"/>
> >        
> >        </condition>
> >  
> >  </extension>
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org



From msc at freeswitch.org  Wed Jun  8 20:03:39 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 8 Jun 2011 09:03:39 -0700
Subject: [Freeswitch-users] An extension with a javascript action can
 handle multiple concurrent calls?
In-Reply-To: <201106081140.45399.sos@sokhapkin.dyndns.org>
References: <33095823FD21DF429B481B5163264B7950AC5864D7@VMBX102.ihostexchange.net>
	<BANLkTi=_OxE7kT7gdSEKK2TCjWQG74CSdw@mail.gmail.com>
	<201106081140.45399.sos@sokhapkin.dyndns.org>
Message-ID: <BANLkTi=oHv-K2rZSS=nJU1cshPZGODq8CA@mail.gmail.com>

I've never ever had a Lua script crash, but I don't use LuaSQL or even the
dbh. However, Chad Phillips (IRC: hunmonk) uses Lua and dbh for the
JesterMail.

-MC

On Wed, Jun 8, 2011 at 8:40 AM, Sergey Okhapkin <sos at sokhapkin.dyndns.org>wrote:

> To me lua frequently crashes when multiple channels call lua interpreter at
> the same time. Even with trivial lua script which calls single select
> statement using dbh.
>
> On Wednesday 08 June 2011, Michael Collins wrote:
> > Yes, you'll have one js instance for each call. JS does not scale very
> > well, but if you want the ability to scale a lot then you should use Lua.
> >
> > -MC
> >
> > On Wed, Jun 8, 2011 at 8:30 AM, Yungwei Chen <yungwei at resolvity.com>
> wrote:
> > > Hi,
> > >
> > > I'm wondering if the following extension can handle multiple concurrent
> > > calls. Thanks.
> > >
> > >  <extension name="test">
> > >
> > >        <condition field="destination_number" expression="^9912$">
> > >
> > >                <action application="answer"/>
> > >                <action application="javascript" data="test.js"/>
> > >
> > >        </condition>
> > >
> > >  </extension>
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Wed Jun  8 20:10:36 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 8 Jun 2011 09:10:36 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <BANLkTimUnFWvRoreh5F0mS=tGz2NptWJ0A@mail.gmail.com>

Hello gang!

Today's agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2011_06_08

We have a special guest from George Washington University today: Robert
Daniel. We are looking forward to hearing from him about how FreeSWITCH is
being used in his telecommunications classes.

-MC
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From david.ponzone at ipeva.fr  Wed Jun  8 20:44:26 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 8 Jun 2011 18:44:26 +0200
Subject: [Freeswitch-users] FreeTDM - destination_number empty on
	inbound call
In-Reply-To: <BANLkTimCRbEutxEf2vqt6OrbyVwn0XhHdw@mail.gmail.com>
References: <BANLkTimCRbEutxEf2vqt6OrbyVwn0XhHdw@mail.gmail.com>
Message-ID: <8A8F0584-3226-48F1-989F-83CDBE80AA4D@ipeva.fr>

Christian,

well you should first tell us what kinda of ISDN you ordered from your local telco.
And also, check if the card/drivers are configured with the right ISDN type.
That could explain why you don't get the DID.

I suppose there are other possible reasons:
How many DID do you have ?
If only one, I suppose some telcos may send you the call without anything, as you dont need the info anyway. That would be ugly, but some telcos do ugly things in some places.

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

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

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




Le 08/06/2011 ? 16:34, Christian Benke a ?crit :

> Hi!
> 
> My FS is connected to a plain BRI-line via a HFC-S PCI-card. On
> inbound calls i don't receive a dialed number. Please see the debug at
> http://pastebin.freeswitch.org/16456 for details.
> Do i have to fetch the DID from a variable before i can handle the
> call or is there some other problem, which i assume?
> 
> Outbound calls work just fine btw.
> 
> Best regards,
> Christian
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From yungwei at resolvity.com  Wed Jun  8 21:43:50 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Wed, 8 Jun 2011 13:43:50 -0400
Subject: [Freeswitch-users] Passing a map to session.setVariable()
Message-ID: <33095823FD21DF429B481B5163264B7950AC586567@VMBX102.ihostexchange.net>

Hi,

In javascript, is passing a map or array to session.setVariable() supported? Thanks.


From benkokakao at gmail.com  Wed Jun  8 21:44:44 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Wed, 8 Jun 2011 19:44:44 +0200
Subject: [Freeswitch-users] FreeTDM - destination_number empty on
	inbound call
In-Reply-To: <8A8F0584-3226-48F1-989F-83CDBE80AA4D@ipeva.fr>
References: <BANLkTimCRbEutxEf2vqt6OrbyVwn0XhHdw@mail.gmail.com>
	<8A8F0584-3226-48F1-989F-83CDBE80AA4D@ipeva.fr>
Message-ID: <BANLkTimKEWSvGGsB43Tsy5syNnpN=LnyrA@mail.gmail.com>

On 8 June 2011 18:44, David Ponzone <david.ponzone at ipeva.fr> wrote:
> Christian,
> well you should first tell us what kinda of ISDN you ordered from your local
> telco.
> And also, check if the card/drivers are configured with the right ISDN type.
> That could explain why you don't get the DID.
> I suppose there are other possible reasons:
> How many DID do you have ?
> If only one, I suppose some telcos may send you the call without anything,
> as you dont need the info anyway. That would be ugly, but some telcos do
> ugly things in some places.

Well the line is in our lab since ages, so i can't really tell you
what "exactly" has been ordered back then :-)
It's an EuroISDN-BRI without a MSN but an unbounded number of
DIDs(Fixed 7-digit Headnumber plus up to 5 digit extensions - which is
basically a standard BRI-line in Austria).

As the outbound calls work fine(headnumber+extensions are delivered) i
assume it's not exactly an configuration issue.
According to my colleagues the DID has always been sent(With proprietary PBX).

A few harder facts - my configuration:
http://pastebin.com/Kc4BPL9E
This in old paste of a previous problem, but the configuration is
still the same. If i can provide you with more precise information,
please let me know what you need, a ISDN-trace is in the paste in my
previous mail.

I'll run a test on a different box with a Sangoma card tomorrow, i
just hoped someone could give me a jump-start with the information i
posted earlier.

Best,
Christian


From steveayre at gmail.com  Wed Jun  8 22:01:08 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 8 Jun 2011 19:01:08 +0100
Subject: [Freeswitch-users] Passing a map to session.setVariable()
In-Reply-To: <33095823FD21DF429B481B5163264B7950AC586567@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950AC586567@VMBX102.ihostexchange.net>
Message-ID: <BANLkTikzZo24EM6==JYMbgnBkoJGZn2W+w@mail.gmail.com>

Variables can only store strings. You'd want to serialise the map/array to a
string and storing that. What do you need to store?

-Steve


On 8 June 2011 18:43, Yungwei Chen <yungwei at resolvity.com> wrote:

> Hi,
>
> In javascript, is passing a map or array to session.setVariable()
> supported? Thanks.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anton.vazir at gmail.com  Wed Jun  8 22:09:27 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Wed, 8 Jun 2011 23:09:27 +0500
Subject: [Freeswitch-users] LuaSQL, Curl, mod_skypopen,
 mod_dingaling....Can't get 'em to work!
In-Reply-To: <BANLkTi=xk8esC+xzo7=5_t5H=BqcXe5g6Q@mail.gmail.com>
References: <31778696.post@talk.nabble.com>
	<BANLkTi=xk8esC+xzo7=5_t5H=BqcXe5g6Q@mail.gmail.com>
Message-ID: <BANLkTikAdhbMK=PVWj5Fvj7gBWu_LhUz8Q@mail.gmail.com>

all works for me.

2011/6/8 Giovanni Maruzzelli <gmaruzz at gmail.com>:
> First read all the related docs in the wiki: http://wiki.freeswitch.org
> Then, search the mailing list archive: lists.freeswitch.org
>
> After that, if you have not found the answers to your doubts, please
> write here what is your specific problem, with all the details that
> can give us the precise idea of why you are in a situation that is not
> documented.
>
> -giovanni
>
> On Wed, Jun 8, 2011 at 2:04 AM, Mint.Nick <mint.office.nick at gmail.com> wrote:
>>
>> Hi all,
>> I've noticed there isn't much traffic on this forum anymore. I'm having
>> issues with the things listed in the subj. Is there anyone out there who
>> might be willing to help a guy figure one or more of these out?
>>
>> Thanks
>> -Nick
>> --
>> View this message in context: http://old.nabble.com/LuaSQL%2C-Curl%2C-mod_skypopen%2C-mod_dingaling....Can%27t-get-%27em-to-work%21-tp31778696p31778696.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From yungwei at resolvity.com  Wed Jun  8 22:18:13 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Wed, 8 Jun 2011 14:18:13 -0400
Subject: [Freeswitch-users] Enumerate variables in a session
Message-ID: <33095823FD21DF429B481B5163264B7950AC586589@VMBX102.ihostexchange.net>

Hi, 

I'm wondering if there's way to enumerate the list of variable names in a session in javascript. Thanks.


From stkn at freeswitch.org  Wed Jun  8 22:43:49 2011
From: stkn at freeswitch.org (Stefan Knoblich)
Date: Wed, 8 Jun 2011 20:43:49 +0200
Subject: [Freeswitch-users] FreeTDM - destination_number empty on
	inbound call
In-Reply-To: <BANLkTimCRbEutxEf2vqt6OrbyVwn0XhHdw@mail.gmail.com>
References: <BANLkTimCRbEutxEf2vqt6OrbyVwn0XhHdw@mail.gmail.com>
Message-ID: <201106082043.54622.stkn@freeswitch.org>

Am Wednesday 08 June 2011 schrieb Christian Benke:
> My FS is connected to a plain BRI-line via a HFC-S PCI-card. On
> inbound calls i don't receive a dialed number. Please see the debug at
> http://pastebin.freeswitch.org/16456 for details.
> Do i have to fetch the DID from a variable before i can handle the
> call or is there some other problem, which i assume?

That SETUP message at the top isn't valid, AFAIK.
There are two 'Calling Number' IEs (0x6c) and the second
one should have been a 'Called Number' IE (0x70) instead
('190341' is the number you dialed, right?).

-- 
-------------------------------------------------------------------------------
Stefan Knoblich                        | Web:   http://www.axsentis.de/
axsentis GmbH                          |        http://oss.axsentis.de/
Eupener Str. 74, 50933 Koeln, Germany  |	
Amtsgericht Koeln: HR B 56238          | Email: s.knoblich at axsentis.de
UST-ID: DE244977565                    | JID:   s.knoblich at jabber.axsentis.de
-------------------------------------------------------------------------------
Web:   http://stkn.techmage.de/
Email: stkn at freeswitch.org
IRC:   #freeswitch-de @ irc.freenode.net
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From eric at loopfx.com  Wed Jun  8 23:11:34 2011
From: eric at loopfx.com (Eric Beard)
Date: Wed, 8 Jun 2011 15:11:34 -0400
Subject: [Freeswitch-users] IP Whitelist
Message-ID: <D4A666272A214C4DBBABFE569CA34CA401DFFC7617@doctorbob.loop.local>

It seems I misunderstand the purpose of the acl.conf.xml file.

What I want to do is create an IP whitelist, so only the IPs I designate get a response from FreeSwitch.  I'd like to do this with FreeSwitch rather than a firewall.

I have this in acl.conf.xml:

    <list name="domains" default="deny">
      <!-- domain= is special it scans the domain from the directory to build the ACL -->
      <node type="allow" domain="$${domain}"/>
      <!-- use cidr= if you wish to allow ip ranges to this domains acl. -->
      <node type="allow" cidr="10.1.0.0/24"/>

      <!-- Broadvox DID -->
      <node type="allow" cidr="209.249.3.74/32"/>
    </list>

I was assuming that this would only allow traffic from my local network, 10.1.0.0, and from the single IP 209.249.3.74

But while watching sip traffic, I saw an OPTIONS request from a different IP (sipvicious scan).  Freeswitch happily responded to the OPTIONS with an OK.

How can I configure it so that it ignores requests that are not on my whitelist?

Thanks!

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com<mailto:eric at loopfx.com>

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From kris at livecall.com  Wed Jun  8 23:12:06 2011
From: kris at livecall.com (Kris)
Date: Wed, 8 Jun 2011 12:12:06 -0700
Subject: [Freeswitch-users] Passing a map to session.setVariable()
References: <33095823FD21DF429B481B5163264B7950AC586567@VMBX102.ihostexchange.net>
	<BANLkTikzZo24EM6==JYMbgnBkoJGZn2W+w@mail.gmail.com>
Message-ID: <BF917D29FCDC4931915EC8058F1AE5EC@stor1>

Why not pass the file name?

----- Original Message ----- 
From: "Steven Ayre" <steveayre at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, June 08, 2011 11:01 AM
Subject: Re: [Freeswitch-users] Passing a map to session.setVariable()


> Variables can only store strings. You'd want to serialise the map/array to 
> a
> string and storing that. What do you need to store?
>
> -Steve
>
>
> On 8 June 2011 18:43, Yungwei Chen <yungwei at resolvity.com> wrote:
>
>> Hi,
>>
>> In javascript, is passing a map or array to session.setVariable()
>> supported? Thanks.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 



From a.luppi at seletech.com  Wed Jun  8 23:14:23 2011
From: a.luppi at seletech.com (Alessandro)
Date: Wed, 08 Jun 2011 21:14:23 +0200
Subject: [Freeswitch-users] Response status from client
Message-ID: <4DEFCA0F.7090808@seletech.com>

Hi,

I have two questions about FS:

Question 1:
i'm developing a custom client sip with pjsip. This client when receive 
a call that can't be accepted respond with status 603. I think that 
freeswitch filter this status.
This is an example of desired behaviour:
1000 at localnet_ip                      FS(ip:localnet_ip)               
                        1001 at localnet_ip

INVITE ---------->
                                      INVITE --------------->
<-------------- trying<--------------------trying
<-------------- 603<------------------  603
ACK ------------>                     ACK------------------>

The current behaviour of FS is:

1000 at localnet_ip                      FS(ip:localnet_ip)               
                        1001 at localnet_ip

INVITE ---------->
                                      INVITE --------------->
<-------------- trying<--------------------trying
                                      <------------------  603

<-------------- 200

ACK------------------>
<--------------------BYE

I'd like to avoid the current behaviour. It's possible a kind of message status path trough?
If the called party terminate the call before answering, FS send always to the calling partner 200 and BYE.
First thought was related to the voice-mail. Now voice-mail is disabled but the behaviour is the same.

Question:2
It's possible a custom Header pass trough in status response like trying or session in progress?
I'm able to use custom header only on invite adding to invite a header with name like X-myheader.
Any suggestion?

Thanks
Good Evening

Alessandro

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

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From steveayre at gmail.com  Wed Jun  8 23:44:22 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 8 Jun 2011 20:44:22 +0100
Subject: [Freeswitch-users] IP Whitelist
In-Reply-To: <D4A666272A214C4DBBABFE569CA34CA401DFFC7617@doctorbob.loop.local>
References: <D4A666272A214C4DBBABFE569CA34CA401DFFC7617@doctorbob.loop.local>
Message-ID: <BANLkTi=Monp1fMy0E3Bbq9P5oc-_0X7YFA@mail.gmail.com>

ACLs control registrations and calls, not options requests.

You'd be best off blocking sipvicious with this iptables entry:

iptables -I INPUT -j DROP -p udp --dport 5060 -m string --string
"friendly-scanner" --algo bm


-Steve


On 8 June 2011 20:11, Eric Beard <eric at loopfx.com> wrote:

> It seems I misunderstand the purpose of the acl.conf.xml file.
>
>
>
> What I want to do is create an IP whitelist, so only the IPs I designate
> get a response from FreeSwitch.  I?d like to do this with FreeSwitch rather
> than a firewall.
>
>
>
> I have this in acl.conf.xml:
>
>
>
>     <list name="domains" default="deny">
>
>       <!-- domain= is special it scans the domain from the directory to
> build the ACL -->
>
>       <node type="allow" domain="$${domain}"/>
>
>       <!-- use cidr= if you wish to allow ip ranges to this domains acl.
> -->
>
>       <node type="allow" cidr="10.1.0.0/24"/>
>
>
>
>       <!-- Broadvox DID -->
>
>       <node type="allow" cidr="209.249.3.74/32"/>
>
>     </list>
>
>
>
> I was assuming that this would only allow traffic from my local network,
> 10.1.0.0, and from the single IP 209.249.3.74
>
>
>
> But while watching sip traffic, I saw an OPTIONS request from a different
> IP (sipvicious scan).  Freeswitch happily responded to the OPTIONS with an
> OK.
>
>
>
> How can I configure it so that it ignores requests that are not on my
> whitelist?
>
>
>
> Thanks!
>
>
>
> -----------------------
>
> *Eric Z. Beard, CTO*
>
> Loop LLC
>
> w (877) 850-2010 x9249
>
> m (727) 776-2768
>
> eric at loopfx.com
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Wed Jun  8 23:47:52 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 8 Jun 2011 20:47:52 +0100
Subject: [Freeswitch-users] LuaSQL, Curl, mod_skypopen,
 mod_dingaling....Can't get 'em to work!
In-Reply-To: <31778696.post@talk.nabble.com>
References: <31778696.post@talk.nabble.com>
Message-ID: <BANLkTinZUx2MDf63cw6bwKOdGaLR0UL4VA@mail.gmail.com>

Rather than using LuaSQL you might find you have more success with the
freeswitch.Dbh class that gives you access to FreeSWITCH's ODBC support from
within lua:
http://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh

If you still need help after reading the documentation, there are almost
always people on the IRC channel that can help you figure things out
(#freeswitch on freenode). This mailing list does see plenty of traffic
though too, although you might not get as fast an answer.

-Steve


On 8 June 2011 01:04, Mint.Nick <mint.office.nick at gmail.com> wrote:

>
> Hi all,
> I've noticed there isn't much traffic on this forum anymore. I'm having
> issues with the things listed in the subj. Is there anyone out there who
> might be willing to help a guy figure one or more of these out?
>
> Thanks
> -Nick
> --
> View this message in context:
> http://old.nabble.com/LuaSQL%2C-Curl%2C-mod_skypopen%2C-mod_dingaling....Can%27t-get-%27em-to-work%21-tp31778696p31778696.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Wed Jun  8 23:55:20 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 8 Jun 2011 20:55:20 +0100
Subject: [Freeswitch-users] Response status from client
In-Reply-To: <4DEFCA0F.7090808@seletech.com>
References: <4DEFCA0F.7090808@seletech.com>
Message-ID: <BANLkTim5=tvyxYJgq3AhtfMd2=NHWvQx+Q@mail.gmail.com>

>
> Question 1:
> i'm developing a custom client sip with pjsip. This client when receive a
> call that can't be accepted respond with status 603. I think that freeswitch
> filter this status.
>

603 gets treated fine for me. I think we need to see more information - can
you put a debug level log of the call with siptrace enabled (sofia global
siptrace on) on pastebin (http://pastebin.freeswitch.org/) and then post the
url here?

Chances are you're doing something in the dialplan that's answering the
call, either before or after the failed bleg.

You said you had voicemail before... you can't send 603 back to the client
and continue to voicemail because the 603 terminates the call.


> Question:2
>
It's possible a custom Header pass trough in status response like trying or
> session in progress? I'm able to use custom header only on invite adding to
> invite a header with name like X-myheader. Any suggestion?
>

Yes, you can for 180/183, with the sip_ph_X- prefix. That puts the header on
any provisional response.
http://wiki.freeswitch.org/wiki/Sofia#Adding_Response_Headers

For example:
<action application="set" data="sip_p_X-myheader=value"/>

AFAIK you won't be able to do the same for a 100 Trying since Sofia doesn't
let FS do any handling at the required point. But have a try anyway just to
be sure.

-Steve


On 8 June 2011 20:14, Alessandro <a.luppi at seletech.com> wrote:

>  Hi,
>
> I have two questions about FS:
>
> Question 1:
> i'm developing a custom client sip with pjsip. This client when receive a
> call that can't be accepted respond with status 603. I think that freeswitch
> filter this status.
> This is an example of desired behaviour:
>  1000 at localnet_ip                      FS(ip:localnet_ip)
>                        1001 at localnet_ip
>
>  INVITE ---------->
>                                      INVITE --------------->
> <-------------- trying                <--------------------trying
> <-------------- 603                  <------------------  603
> ACK ------------>                    ACK------------------>
>
>
>
> The current behaviour of FS is:
>
>  1000 at localnet_ip                      FS(ip:localnet_ip)
>                        1001 at localnet_ip
>
>  INVITE ---------->
>                                      INVITE --------------->
> <-------------- trying                <--------------------trying
>                                      <------------------  603
>
> <-------------- 200
>
> ACK------------------> <--------------------BYE I'd like to avoid the
> current behaviour. It's possible a kind of message status path trough? If
> the called party terminate the call before answering, FS send always to the
> calling partner 200 and BYE. First thought was related to the voice-mail.
> Now voice-mail is disabled but the behaviour is the same. Question:2 It's
> possible a custom Header pass trough in status response like trying or
> session in progress? I'm able to use custom header only on invite adding to
> invite a header with name like X-myheader. Any suggestion? Thanks Good
> Evening
>
>  Alessandro
>
> --
> Ing. Alessandro Luppi
> Software development
> Seletech srl
> Via Collodi 8, 20052 Monza (MI) - Italy
> Tel: +39.039.5962000 - Fax: +39.039.9716905
> email: a.luppi at seletech.com - Web: www.seletech.com   or   www.seletech.eu
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Wed Jun  8 23:56:10 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 8 Jun 2011 20:56:10 +0100
Subject: [Freeswitch-users] Passing a map to session.setVariable()
In-Reply-To: <BF917D29FCDC4931915EC8058F1AE5EC@stor1>
References: <33095823FD21DF429B481B5163264B7950AC586567@VMBX102.ihostexchange.net>
	<BANLkTikzZo24EM6==JYMbgnBkoJGZn2W+w@mail.gmail.com>
	<BF917D29FCDC4931915EC8058F1AE5EC@stor1>
Message-ID: <BANLkTimAh_1AdxLUSEjS5Wm+CK9_c6G5ww@mail.gmail.com>

Well you can, but he never said he had a filename to pass...


On 8 June 2011 20:12, Kris <kris at livecall.com> wrote:

> Why not pass the file name?
>
> ----- Original Message -----
> From: "Steven Ayre" <steveayre at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Wednesday, June 08, 2011 11:01 AM
> Subject: Re: [Freeswitch-users] Passing a map to session.setVariable()
>
>
> > Variables can only store strings. You'd want to serialise the map/array
> to
> > a
> > string and storing that. What do you need to store?
> >
> > -Steve
> >
> >
> > On 8 June 2011 18:43, Yungwei Chen <yungwei at resolvity.com> wrote:
> >
> >> Hi,
> >>
> >> In javascript, is passing a map or array to session.setVariable()
> >> supported? Thanks.
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From eric at loopfx.com  Thu Jun  9 00:08:00 2011
From: eric at loopfx.com (Eric Beard)
Date: Wed, 8 Jun 2011 16:08:00 -0400
Subject: [Freeswitch-users] IP Whitelist
In-Reply-To: <BANLkTi=Monp1fMy0E3Bbq9P5oc-_0X7YFA@mail.gmail.com>
References: <D4A666272A214C4DBBABFE569CA34CA401DFFC7617@doctorbob.loop.local>
	<BANLkTi=Monp1fMy0E3Bbq9P5oc-_0X7YFA@mail.gmail.com>
Message-ID: <D4A666272A214C4DBBABFE569CA34CA401DFFC7691@doctorbob.loop.local>

Is it actually supposed to prevent calls from any IP's not in acl.conf.xml?  With the settings I listed below, I can still make calls from different IP's to any number, which effectively makes my server an open relay.  Do those settings only work if you are authenticating callers?

It seems like I'm going to have to set up IP-specific firewall rules for each SIP port, to allow only traffic from my gateways.

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com<mailto:eric at loopfx.com>

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: Wednesday, June 08, 2011 3:44 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] IP Whitelist

ACLs control registrations and calls, not options requests.

You'd be best off blocking sipvicious with this iptables entry:

iptables -I INPUT -j DROP -p udp --dport 5060 -m string --string "friendly-scanner" --algo bm

-Steve

On 8 June 2011 20:11, Eric Beard <eric at loopfx.com<mailto:eric at loopfx.com>> wrote:
It seems I misunderstand the purpose of the acl.conf.xml file.

What I want to do is create an IP whitelist, so only the IPs I designate get a response from FreeSwitch.  I'd like to do this with FreeSwitch rather than a firewall.

I have this in acl.conf.xml:

    <list name="domains" default="deny">
      <!-- domain= is special it scans the domain from the directory to build the ACL -->
      <node type="allow" domain="$${domain}"/>
      <!-- use cidr= if you wish to allow ip ranges to this domains acl. -->
      <node type="allow" cidr="10.1.0.0/24<http://10.1.0.0/24>"/>

      <!-- Broadvox DID -->
      <node type="allow" cidr="209.249.3.74/32<http://209.249.3.74/32>"/>
    </list>

I was assuming that this would only allow traffic from my local network, 10.1.0.0, and from the single IP 209.249.3.74

But while watching sip traffic, I saw an OPTIONS request from a different IP (sipvicious scan).  Freeswitch happily responded to the OPTIONS with an OK.

How can I configure it so that it ignores requests that are not on my whitelist?

Thanks!

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com<mailto:eric at loopfx.com>


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

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From eric at loopfx.com  Thu Jun  9 00:24:46 2011
From: eric at loopfx.com (Eric Beard)
Date: Wed, 8 Jun 2011 16:24:46 -0400
Subject: [Freeswitch-users] IP Whitelist
In-Reply-To: <D4A666272A214C4DBBABFE569CA34CA401DFFC7691@doctorbob.loop.local>
References: <D4A666272A214C4DBBABFE569CA34CA401DFFC7617@doctorbob.loop.local>
	<BANLkTi=Monp1fMy0E3Bbq9P5oc-_0X7YFA@mail.gmail.com>
	<D4A666272A214C4DBBABFE569CA34CA401DFFC7691@doctorbob.loop.local>
Message-ID: <D4A666272A214C4DBBABFE569CA34CA401DFFC76C4@doctorbob.loop.local>

I found what I was missing.  Since I don't authenticate callers, I use the external profile for everything, even calls from my LAN.  I had never copied this setting from internal.xml:

<param name="apply-inbound-acl" value="domains"/>

I put that in sip_profiles/external.xml.

Now FreeSwitch sends 403 to any IPs not in acl.conf.xml.

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com<mailto:eric at loopfx.com>

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Eric Beard
Sent: Wednesday, June 08, 2011 4:08 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] IP Whitelist

Is it actually supposed to prevent calls from any IP's not in acl.conf.xml?  With the settings I listed below, I can still make calls from different IP's to any number, which effectively makes my server an open relay.  Do those settings only work if you are authenticating callers?

It seems like I'm going to have to set up IP-specific firewall rules for each SIP port, to allow only traffic from my gateways.

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com<mailto:eric at loopfx.com>

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: Wednesday, June 08, 2011 3:44 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] IP Whitelist

ACLs control registrations and calls, not options requests.

You'd be best off blocking sipvicious with this iptables entry:

iptables -I INPUT -j DROP -p udp --dport 5060 -m string --string "friendly-scanner" --algo bm

-Steve
On 8 June 2011 20:11, Eric Beard <eric at loopfx.com<mailto:eric at loopfx.com>> wrote:
It seems I misunderstand the purpose of the acl.conf.xml file.

What I want to do is create an IP whitelist, so only the IPs I designate get a response from FreeSwitch.  I'd like to do this with FreeSwitch rather than a firewall.

I have this in acl.conf.xml:

    <list name="domains" default="deny">
      <!-- domain= is special it scans the domain from the directory to build the ACL -->
      <node type="allow" domain="$${domain}"/>
      <!-- use cidr= if you wish to allow ip ranges to this domains acl. -->
      <node type="allow" cidr="10.1.0.0/24<http://10.1.0.0/24>"/>

      <!-- Broadvox DID -->
      <node type="allow" cidr="209.249.3.74/32<http://209.249.3.74/32>"/>
    </list>

I was assuming that this would only allow traffic from my local network, 10.1.0.0, and from the single IP 209.249.3.74

But while watching sip traffic, I saw an OPTIONS request from a different IP (sipvicious scan).  Freeswitch happily responded to the OPTIONS with an OK.

How can I configure it so that it ignores requests that are not on my whitelist?

Thanks!

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com<mailto:eric at loopfx.com>


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

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From kheimerl at cs.berkeley.edu  Thu Jun  9 01:32:17 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Wed, 8 Jun 2011 14:32:17 -0700
Subject: [Freeswitch-users] Sofia invite issues
Message-ID: <BANLkTikWTD7bT+kiMdDk1VTOU2XsPU0rcQ@mail.gmail.com>

Hello Freeswitch-users!

I'm currently writing a SIP middlebox that intercepts sip messages and
changes their username. The reasons for this are complicated and
somewhat outside of the scope of this discussion. However, I've come
upon a very strange issue: when making a phone-to-phone call across
freeswitch; it is connecting the wrong user. I'm fairly well convinced
this is a bug, but I thought I'd send the issue here and see if it's
anything I'm obviously doing wrong.

Basically, FS sees a SIP message from a registered UA (call it A)
inviting another registered user to a call (B). This is acked
correctly (a TRYING message). FS then responds by inviting B into a
call with some OTHER user C. C is not mentioned at any point in the
initial SIP messages (as verified by wireshark). Even more strangely,
if I remove user C from FS (by removing their config file) FS responds
to the invite by inviting B into a call with itself. In each case, the
appropriate from header in the invite should be the original caller A.
 It's worth noting that I am not monkeying with the RTP packets at
all, but my understanding is that SIP signalling shouldn't be affected
by that.

My general guess is that I'm messing up the naming somehow, and FS is
running an algorithm to guess at who the call originator is. However,
the naming must be roughly correct; Asterisk is able to handle this
call just fine. I've included a sip trace of the second situation (A
calls B, FS invites B to a call with itself)

For the record A:1300
B:1301
C:IMSI641104878332498

REGISTER MESSAGE (Which works fine)

REGISTER sip:192.168.1.144 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
From: 1300 <sip:1300 at 192.168.1.144>;tag=ftoui
To: 1300 <sip:1300 at 192.168.1.144>
Call-ID: 1032827938 at 192.168.1.144
CSeq: 91 REGISTER
Contact: <sip:1300 at 192.168.1.144:5063>;expires=7200
Max-Forwards: 70
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
From: 1300 <sip:1300 at 192.168.1.144>;tag=ftoui
To: 1300 <sip:1300 at 192.168.1.144>;tag=jU64NXypQc57F
Call-ID: 1032827938 at 192.168.1.144
CSeq: 91 REGISTER
Contact: <sip:1300 at 192.168.1.144:5063>;expires=7200
Date: Wed, 08 Jun 2011 21:02:04 GMT
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-71d64e6 2011-06-07 22-30-16 -0700
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Length: 0

SIP TRACE (A -> B, B -> B response)

INITIAL INVITE
INVITE sip:1301 at 192.168.1.144 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK77974
From: 1300 <sip:1300 at 192.168.1.144>;tag=bgdqx
To: <sip:1301 at 192.168.1.144>
Call-ID: 1817795092 at 192.168.1.144
CSeq: 485 INVITE
Contact: <sip:1300 at 192.168.1.144:5063>;expires=3600
Content-Type: application/sdp
Max-Forwards: 70
Content-Length: 143

INVITE ACK
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK77974
From: 1300 <sip:1300 at 192.168.1.144>;tag=bgdqx
To: <sip:1301 at 192.168.1.144>
Call-ID: 1817795092 at 192.168.1.144
CSeq: 485 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-71d64e6 2011-06-07 22-30-16 -0700
Content-Length: 0

FS INVITE (Note the from address being wrong)
INVITE sip:1301 at 192.168.1.144:5063 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.144;rport;branch=z9hG4bKeeQvcyZ70SDUg
Max-Forwards: 69
From: "1301" <sip:1301 at 192.168.1.144>;tag=rHyS0Z3B61arN
To: <sip:1301 at 192.168.1.144:5063>
Call-ID: 84df47f9-0cb7-122f-13b5-5cff350d9de5
CSeq: 13447852 INVITE
Contact: <sip:mod_sofia at 192.168.1.144:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-71d64e6 2011-06-07 22-30-16 -0700
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
sla, include-session-description, presence.winfo, message-summary,
refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 315
X-FS-Support: update_display
Remote-Party-ID: "1301"
<sip:1301 at 192.168.1.144>;party=calling;screen=yes;privacy=off

USER CONFIGURATIONS

1300.xml
<include>
  <user id="1300" cidr="192.168.1.0/24,127.0.0.1/32">
    <params>
      <param name="accept-blind-reg" value="true"/>
      <param name="accept-blind-auth" value="true"/>
      <param name="vm-password" value="1000"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="1300"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="1300"/>
      <variable name="effective_caller_id_number" value="1300"/>
      <variable name="outbound_caller_id_name" value="1300"/>
      <variable name="outbound_caller_id_number" value="1300"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>

1301.xml
<include>
  <user id="1301" cidr="192.168.1.0/24,127.0.0.1/32">
    <params>
      <param name="accept-blind-reg" value="true"/>
      <param name="accept-blind-auth" value="true"/>
      <param name="vm-password" value="1000"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="1301"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="1301"/>
      <variable name="effective_caller_id_number" value="1301"/>
      <variable name="outbound_caller_id_name" value="1301"/>
      <variable name="outbound_caller_id_number" value="1301"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>

IMSI641104878332498.xml
<include>
  <user id="IMSI641104878332498" cidr="192.168.1.0/24,127.0.0.1/32"
number-alias="1302">
    <params>
      <param name="accept-blind-reg" value="true"/>
      <param name="accept-blind-auth" value="true"/>
      <param name="vm-password" value="1000"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="1302"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="IMSI641104878332498"/>
      <variable name="effective_caller_id_number" value="1302"/>
      <variable name="outbound_caller_id_name" value="IMSI641104878332498"/>
      <variable name="outbound_caller_id_number" value="1302"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>

DIALPLAN

    <extension name="Local Numbers">
      <condition field="destination_number" expression="^130[0-9]$">
	<!--<action application="bridge"
data="user/${user_data(${destination_number}@${domain_name} attr
id)}"/> -->
	<action application="bridge" data="user/${destination_number}"/>
      </condition>
    </extension>


From msc at freeswitch.org  Thu Jun  9 01:51:21 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 8 Jun 2011 14:51:21 -0700
Subject: [Freeswitch-users] Passing a map to session.setVariable()
In-Reply-To: <BANLkTimAh_1AdxLUSEjS5Wm+CK9_c6G5ww@mail.gmail.com>
References: <33095823FD21DF429B481B5163264B7950AC586567@VMBX102.ihostexchange.net>
	<BANLkTikzZo24EM6==JYMbgnBkoJGZn2W+w@mail.gmail.com>
	<BF917D29FCDC4931915EC8058F1AE5EC@stor1>
	<BANLkTimAh_1AdxLUSEjS5Wm+CK9_c6G5ww@mail.gmail.com>
Message-ID: <BANLkTi=yHpgHaQCa4rDvCk28OXRwvQOqwA@mail.gmail.com>

FYI, in the latest and greatest version of FreeSWITCH we *do* have arrays! I
haven't had a chance to document it yet, but you can check it out:

http://fisheye.freeswitch.org/changelog/freeswitch.git/?cs=c1c759526d1ccf7cca8afcd139751133d382d34e
and
http://fisheye.freeswitch.org/changelog/freeswitch.git/?cs=860d2a6c12cb5dc1c4e63ecc92b7610bcacf5902

In fact, if anyone knows about these and would like to help me get them
wikified then please let me know.

-MC


On Wed, Jun 8, 2011 at 12:56 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Well you can, but he never said he had a filename to pass...
>
>
>
> On 8 June 2011 20:12, Kris <kris at livecall.com> wrote:
>
>> Why not pass the file name?
>>
>> ----- Original Message -----
>> From: "Steven Ayre" <steveayre at gmail.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Wednesday, June 08, 2011 11:01 AM
>> Subject: Re: [Freeswitch-users] Passing a map to session.setVariable()
>>
>>
>> > Variables can only store strings. You'd want to serialise the map/array
>> to
>> > a
>> > string and storing that. What do you need to store?
>> >
>> > -Steve
>> >
>> >
>> > On 8 June 2011 18:43, Yungwei Chen <yungwei at resolvity.com> wrote:
>> >
>> >> Hi,
>> >>
>> >> In javascript, is passing a map or array to session.setVariable()
>> >> supported? Thanks.
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From potxoka at gmail.com  Thu Jun  9 02:13:14 2011
From: potxoka at gmail.com (Antonio)
Date: Thu, 09 Jun 2011 00:13:14 +0200
Subject: [Freeswitch-users] Proxy traffic security
Message-ID: <4DEFF3FA.5050500@gmail.com>

Hello

I have a FreeSwitch configured as a gateway, the proxy makes the user 
authentication and other functions. All servers have public ip's, and I 
have doubts to security. I had thought to put the proxy ip's in acl. Is 
it safe this scenario? Can it be improved? Should we also put the sip 
providers? Thanks.

Greetings


From sid.kshatriya at gmail.com  Thu Jun  9 07:21:03 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Thu, 9 Jun 2011 08:51:03 +0530
Subject: [Freeswitch-users] FreeTDM - Sangoma B700 - ISDN connection
 questions - UK
In-Reply-To: <4DED4854.6010203@earthspike.net>
References: <4DED4854.6010203@earthspike.net>
Message-ID: <BANLkTimpvqDfhKEbF=AgaVC3fy1oebY84A@mail.gmail.com>

Dear John,

Appreciate the long time you took in putting down your queries. I'd like to
help but unfortunately don't have domain expertise in the areas you're
having problems.

Your mail is more likely to be read if it was shorter. May I suggest you use
http://pastebin.freeswitch.org/ to paste all your debug messages and
configurations.

Please ignore this message if you don't agree with the strategy.

Thanks,

Sidharth

On Tue, Jun 7, 2011 at 3:06 AM, John <freeswitch at earthspike.net> wrote:

> Hello,
>
> I have just set up a FreeSWITCH box with a Sangoma B700 connected to 2 x
> ISDN2e lines (each 2B+D, so 4 voice channels) in UK, so supplied by BT
> Openreach.  There are a number of anomalies that I am trying to solve.
> [One of these is that 1 line is dead, but that is for BT Openreach to
> resolve rather than anyone on this list.]
>
> I have a few questions, but as they are all related, I hope you don't
> mind them in one post.  Some basics first.  The box is an Atom dual-core
> with 2GB of memory and a Sangoma B700 card.  It's built with Ubuntu
> 10.04.2 LTS server 64-bit, patched and up to date, and also runs dhcpd,
> lighttpd and sshd.  I built the Sangoma ISDN libraries and FreeSWITCH
> using the latest git versions I could ('make current' about 2 weeks
> ago).  We have incoming and outgoing calls working, but some incoming
> calls ring in the caller's ear, but nothing appears on the FreeSWITCH
> console, and others fail.  Rebooting the server cures this.  We have had
> problems with lines being disconnected and then reconnected, and it
> seems that FreeTDM/wanrouter/whatever doesn't recognise previously
> disconnected lines coming back into use, because a reboot finds lines
> that were previously reported disconnected ('wanrouter status' or 'ftdm
> list').  We have ongoing problems with one line that is 'disconnected'
> (wanpipe2/wp2) but the queries I am describing below apply equally when
> both lines are connected and working.
>
> 1. There seems to be a lot of ISDN 'chatter' with channels going up and
> down all the time even when the switch is completely idle.  Is this
> normal?  Here is my /log 7 showing two of the cycles (which appear to be
> about 50s apart):
>
> 2011-06-06 22:01:54.992467 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:681
> [SNGISDN Q921] wp1: Protocol: Data Link connection DN(4): Disconnect
> initiated(263)
> 2011-06-06 22:02:29.952464 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:748
> [SNGISDN Q931] s1: Interface: Down(261): Dchan(285)
> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
> [s1c1][1:1] Signalling link status changed to DOWN
> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
> [s1c2][1:2] Signalling link status changed to DOWN
> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
> [s1c3][1:3] Signalling link status changed to DOWN
> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:200
> [s1c1][1:1] Setting availability rate to:5
> 2011-06-06 22:02:29.952464 [DEBUG] mod_freetdm.c:2268 got clear channel
> sig [SIGSTATUS_CHANGED]
> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:186
> [s1c1][1:1] Setting availability rate to:5
> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:200
> [s1c2][1:2] Setting availability rate to:5
> 2011-06-06 22:02:29.952464 [INFO] mod_freetdm.c:2345 1:1 signalling
> changed to :DOWN
> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:186
> [s1c2][1:2] Setting availability rate to:5
> 2011-06-06 22:02:29.952464 [DEBUG] mod_freetdm.c:2268 got clear channel
> sig [SIGSTATUS_CHANGED]
> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:200
> [s1c3][1:3] Setting availability rate to:5
> 2011-06-06 22:02:29.952464 [INFO] mod_freetdm.c:2345 1:2 signalling
> changed to :DOWN
> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:186
> [s1c3][1:3] Setting availability rate to:5
> 2011-06-06 22:02:29.952464 [DEBUG] mod_freetdm.c:2268 got clear channel
> sig [SIGSTATUS_CHANGED]
> 2011-06-06 22:02:29.952464 [INFO] mod_freetdm.c:2345 1:3 signalling
> changed to :DOWN
> 2011-06-06 22:02:29.952464 [INFO] ftmod_sangoma_isdn_stack_rcv.c:631
> Received RESTART CFM (dChan:1 ces:0 type:1)
> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:1179
> Receved RESTART, but Restart Indicator IE not present
> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:681
> [SNGISDN Q921] wp1: Protocol: Data Link connection UP(3): UA frame with
> F-bit = 1(258)
> 2011-06-06 22:02:29.972464 [INFO] ftmod_sangoma_isdn_stack_rcv.c:631
> Received RESTART CFM (dChan:1 ces:0 type:0)
> 2011-06-06 22:02:29.972464 [INFO] ftmod_sangoma_isdn_stack_rcv.c:729
> [SNGISDN Q931] s1: Interface: UP(260): Dchan(285)
> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
> [s1c1][1:1] Signalling link status changed to UP
> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
> [s1c2][1:2] Signalling link status changed to UP
> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
> [s1c3][1:3] Signalling link status changed to UP
> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:200
> [s1c1][1:1] Setting availability rate to:10
> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:186
> [s1c1][1:1] Setting availability rate to:10
> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:200
> [s1c2][1:2] Setting availability rate to:10
> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:186
> [s1c2][1:2] Setting availability rate to:10
> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:200
> [s1c3][1:3] Setting availability rate to:10
> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:186
> [s1c3][1:3] Setting availability rate to:10
> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:1179
> Receved RESTART, but Restart Indicator IE not present
> 2011-06-06 22:02:29.972464 [DEBUG] mod_freetdm.c:2268 got clear channel
> sig [SIGSTATUS_CHANGED]
> 2011-06-06 22:02:29.972464 [INFO] mod_freetdm.c:2345 1:1 signalling
> changed to :UP
> 2011-06-06 22:02:29.972464 [DEBUG] mod_freetdm.c:2268 got clear channel
> sig [SIGSTATUS_CHANGED]
> 2011-06-06 22:02:29.972464 [INFO] mod_freetdm.c:2345 1:2 signalling
> changed to :UP
> 2011-06-06 22:02:29.972464 [DEBUG] mod_freetdm.c:2268 got clear channel
> sig [SIGSTATUS_CHANGED]
> 2011-06-06 22:02:29.972464 [INFO] mod_freetdm.c:2345 1:3 signalling
> changed to :UP
> 2011-06-06 22:03:19.912462 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:681
> [SNGISDN Q921] wp1: Protocol: Data Link connection DN(4): Disconnect
> initiated(263)
> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:681
> [SNGISDN Q921] wp1: Protocol: Data Link connection UP(3): UA frame with
> F-bit = 1(258)
> 2011-06-06 22:03:19.932483 [INFO] ftmod_sangoma_isdn_stack_rcv.c:631
> Received RESTART CFM (dChan:1 ces:0 type:0)
> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:1179
> Receved RESTART, but Restart Indicator IE not present
> 2011-06-06 22:03:19.932483 [INFO] ftmod_sangoma_isdn_stack_rcv.c:729
> [SNGISDN Q931] s1: Interface: UP(260): Dchan(285)
> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
> [s1c1][1:1] Signalling link status changed to UP
> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
> [s1c2][1:2] Signalling link status changed to UP
> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
> [s1c3][1:3] Signalling link status changed to UP
> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:200
> [s1c1][1:1] Setting availability rate to:10
> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:186
> [s1c1][1:1] Setting availability rate to:10
> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:200
> [s1c2][1:2] Setting availability rate to:10
> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:186
> [s1c2][1:2] Setting availability rate to:10
> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:200
> [s1c3][1:3] Setting availability rate to:10
> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:186
> [s1c3][1:3] Setting availability rate to:10
> 2011-06-06 22:03:19.932483 [DEBUG] mod_freetdm.c:2268 got clear channel
> sig [SIGSTATUS_CHANGED]
> 2011-06-06 22:03:19.932483 [INFO] mod_freetdm.c:2345 1:1 signalling
> changed to :UP
> 2011-06-06 22:03:19.932483 [DEBUG] mod_freetdm.c:2268 got clear channel
> sig [SIGSTATUS_CHANGED]
> 2011-06-06 22:03:19.932483 [INFO] mod_freetdm.c:2345 1:2 signalling
> changed to :UP
> 2011-06-06 22:03:19.932483 [DEBUG] mod_freetdm.c:2268 got clear channel
> sig [SIGSTATUS_CHANGED]
> 2011-06-06 22:03:19.932483 [INFO] mod_freetdm.c:2345 1:3 signalling
> changed to :UP
> freeswitch at internal> ftdm list
> +OK
> span: 1 (wp1)
> type: Sangoma (ISDN)
> physical_status: ok
> signaling_status: UP
> chan_count: 3
> dialplan: XML
> context: public
> dial_regex:
> fail_dial_regex:
> hold_music:
> analog_options: none
> +OK
> span: 2 (wp2)
> type: Sangoma (ISDN)
> physical_status: alarmed
> signaling_status: DOWN
> chan_count: 3
> dialplan: XML
> context: public
> dial_regex:
> fail_dial_regex:
> hold_music:
> analog_options: none
> +OK
> span: 3 (FXS)
> type: analog
> physical_status: ok
> signaling_status: UP
> chan_count: 2
> dialplan: XML
> context: default
> dial_regex:
> fail_dial_regex:
> hold_music:
> analog_options: none
>
>
> # wanrouter status
>
> Devices currently active:
>     wanpipe1 wanpipe2 wanpipe3
>
>
> Wanpipe Config:
>
> Device name | Protocol Map | Adapter  | IRQ | Slot/IO | If's | CLK |
> Baud rate |
> wanpipe1    | N/A          | A500/B700| 20  | 0       | 1    | N/A |
> 0         |
> wanpipe2    | N/A          | A500/B700| 20  | 0       | 1    | N/A |
> 0         |
> wanpipe3    | N/A          | A200/A400/B600/B700/B800| 20  | 0       |
> 1    | N/A | 0         |
>
> Wanrouter Status:
>
> Device name | Protocol | Station | Status        |
> wanpipe1    | AFT ISDN | N/A     | Connected     |
> wanpipe2    | AFT ISDN | N/A     | Disconnected  |
> wanpipe3    | A-ANALOG | N/A     | Connected     |
>
>
> # cat /etc/wanpipe/wanrouter.rc
> #!/bin/sh
> # .. comments snipped ...
> ROUTER_BOOT=YES
> WAN_CONF_DIR=/etc/wanpipe
> WAN_INTR_DIR=/etc/wanpipe/interfaces
> WAN_BIN_DIR=/usr/sbin
> WAN_LOG=/var/log/wanrouter
> WAN_LOCK=/var/lock/wanrouter
> WAN_LOCK_DIR=/var/lock
> WAN_IP_FORWARD=NO
> NEW_IF_TYPE=NO
> WAN_LIB_DIR=/etc/wanpipe/lib
> WAN_ADSL_LIST=/etc/wanpipe/wan_adsl.list
> WAN_ANNEXG_LOAD=NO
> WAN_SCTP_LOAD=NO
> WAN_LIP_LOAD=NO
> WAN_DYN_WANCONFIG=NO
> WAN_SCRIPTS_DIR=/etc/wanpipe/scripts
> WAN_FIRMWARE_DIR=/etc/wanpipe/firmware
> WAN_DEVICES_REV_STOP_ORDER=YES
> WAN_DEVICES="wanpipe1 wanpipe2 wanpipe3 "
>
> # cat /etc/wanpipe/wanpipe1.conf
> #================================================
> # WANPIPE1 Configuration File
> #================================================
> #
> # Note: This file was generated automatically
> #       by /usr/local/sbin/setup-sangoma program.
> #
> #       If you want to edit this file, it is
> #       recommended that you use wancfg program
> #       to do so.
> #================================================
> # Sangoma Technologies Inc.
> #================================================
>
>
>
>
> [devices]
> wanpipe1 = WAN_AFT_ISDN_BRI, Comment
>
> [interfaces]
> w1g1 = wanpipe1, , TDM_VOICE_API, Comment
>
> [wanpipe1]
> CARD_TYPE     = AFT
> S514CPU     = A
> CommPort     = PRI
> AUTO_PCISLOT     = NO
> PCISLOT     = 0
> PCIBUS      = 5
> FE_MEDIA    = BRI
> FE_LINE        = 1
> TDMV_LAW    = ALAW
> RM_BRI_CLOCK_MASTER    = NO
> MTU         = 1500
> UDPPORT     = 9000
> TTL        = 255
> IGNORE_FRONT_END = NO
> TDMV_SPAN    = 1
> TE_AIS_MAINTENANCE = NO            #NO: defualt  YES: Start port in AIS
> Blue Alarm and keep line down
>                                 #wanpipemon -i w1g1 -c Ttx_ais_off to
> disable AIS maintenance mode
>                                 #wanpipemon -i w1g1 -c Ttx_ais_on to
> enable AIS maintenance mode
> TDMV_HW_DTMF        = YES        # YES: receive dtmf events from hardware
> TDMV_HW_FAX_DETECT      = YES     # YES: receive fax 1100hz events from
> hardware
> HWEC_OPERATION_MODE     = OCT_NORMAL    # OCT_NORMAL: echo cancelation
> enabled with nlp (default)
>                                         # OCT_SPEECH: improves software
> tone detection by disabling NLP (echo possible)
>                                         # OCT_NO_ECHO:disables echo
> cancelation but allows VQE/tone functions.
> HWEC_DTMF_REMOVAL       = NO    # NO: default  YES: remove dtmf out of
> incoming media (must have hwdtmf enabled)
> HWEC_NOISE_REDUCTION    = NO    # NO: default  YES: reduces noise on the
> line - could break fax
> HWEC_ACUSTIC_ECHO       = NO    # NO: default  YES: enables acustic echo
> cancelation
> HWEC_NLP_DISABLE        = NO    # NO: default  YES: guarantees software
> tone detection (possible echo)
> HWEC_TX_AUTO_GAIN       = 0     # 0: disable   -40-0: default tx audio
> level to be maintained (-20 default)
> HWEC_RX_AUTO_GAIN       = 0     # 0: disable   -40-0: default tx audio
> level to be maintained (-20 default)
> HWEC_TX_GAIN            = 0     # 0: disable   -24-24: db values to be
> applied to tx signal
> HWEC_RX_GAIN            = 0     # 0: disable   -24-24: db values to be
> applied to tx signal
>
> [w1g1]
> ACTIVE_CH    = ALL
> TDMV_HWEC    = YES
> MTU         = 80
>
>  ... and wanpipe2 was also automatically generated so just has '2' for
> FE_LINE and TDMV_SPAN instead of '1'.
>
> # cat /usr/local/freeswitch/conf/freetdm.conf
> [span wanpipe wp1]
> trunk_type => bri
> group=1
> b-channel => 1:1-2
> d-channel => 1:3
>
> [span wanpipe wp2]
> trunk_type => bri
> group=1
> b-channel => 2:1-2
> d-channel => 2:3
>
> [span wanpipe FXS]
> name => freetdm
> trunk_type => fxs
> group => grp2
> fxs-channel => 3:1
>
> trunk_type => fxs
> group => grp2
> fxs-channel => 3:2
>
> # cat /usr/local/freeswitch/conf/autoload_configs/freetdm.conf.xml
> <configuration name="freetdm.conf" description="Freetdm Configuration">
> <settings>
> <param name="debug" value="0"/>
> <!--<param name="hold-music" value="$${moh_uri}"/>-->
> <!--<param name="enable-analog-option" value="call-swap"/>-->
> <!--<param name="enable-analog-option" value="3-way"/>-->
> </settings>
>
> <config_profiles>
>
> <profile name="my_bri_te_1">
> <param name="switchtype" value="euroisdn" />
> <param name="interface" value="cpe"/>
> <param name="facility" value="no" />
> </profile>
>
> </config_profiles>
>
> <sangoma_bri_spans>
>
> <span name="wp1" cfgprofile="my_bri_te_1">
> <param name="dialplan" value="XML"/>
> <param name="context" value="public"/>
> </span>
>
> <span name="wp2" cfgprofile="my_bri_te_1">
> <param name="dialplan" value="XML"/>
> <param name="context" value="public"/>
> </span>
>
> </sangoma_bri_spans>
>
>
> <analog_spans>
>
> <span name="FXS">
> <!--<param name="hold-music" value="$${moh_uri}"/>-->
> <param name="dialplan" value="XML"/>
> <param name="context" value="default"/>
> <!-- regex to stop dialing when it matches -->
> <!--<param name="dial-regex" value="5555"/>-->
> <!-- regex to stop dialing when it does not match -->
> <!--<param name="fail-dial-regex" value="^5"/>-->
> </span>
>
> </analog_spans>
>
> </configuration>
>
> What other config files do I need to supply? I don't want to deluge the
> list any more in my first post!
>
> 2. We have 2 Single Number DDI numbers configured over the 4 channels.
> I want to set up outgoing calls so that they can appear to come from
> either of the two numbers, but at present the outgoing CLI appears to be
> overridden by the telco (BT) to only use one of the numbers.  Has anyone
> got this working in UK, and what is the format for the
> outbound_caller_id_number: last 6 digits or full 11 digits?  I note that
> the inbound called number is only the last 6 digits.
>
> 3. I have built FreeSWITCH from git and installed at /usr/local/... and
> then followed the steps on the Ubuntu page in the Wiki to set up the run
> control scripts, etc, and run FS non-root as freeswitch:daemon.  With
> FreeTDM, I have discovered that the /dev/wan* devices are owned by
> root:root, and so are inaccessible to FS running as non-root.  So for
> now I have added a line in /etc/init.d/freeswitch to 'chgrp freeswitch
> /dev/wan*'.  This is not the most elegant solution, because 'wanrouter
> restart' (which seems to be my best friend at the moment) resets the
> ownership to root:root.  I have tried grepping to see where the mknods
> are for these devices, but have been unsuccessful.  Is there a better
> place to 'permanently' change the device ownership?
>
> Thanks for all the great support I have already got just from editing my
> Wiki User page; this is a friendly group!
>
> John
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From revmichael at bethelightchapel.com  Thu Jun  9 11:00:48 2011
From: revmichael at bethelightchapel.com (RevMichael)
Date: Thu, 9 Jun 2011 00:00:48 -0700 (PDT)
Subject: [Freeswitch-users] Setup help Conference, 3 way call, FIFO, MOH,
 record to Shoutcast
In-Reply-To: <BANLkTikvOPALgLhTEvNdEnafAgD5fmTimw@mail.gmail.com>
References: <20110520081825.150834ml8unxokqo@psychicawakeningschool.com>
	<1305903105560-6386372.post@n2.nabble.com>
	<20110520110517.1159452g1qi9dyyo@psychicawakeningschool.com>
	<25a801cc1701$6740d510$35c27f30$@com>
	<20110520183037.83314gw215dgz5og@psychicawakeningschool.com>
	<BANLkTikvOPALgLhTEvNdEnafAgD5fmTimw@mail.gmail.com>
Message-ID: <1307602848678-6456754.post@n2.nabble.com>

Conference works great BUT!

I have it setup in the dialplan:

< action application="record"
data="shout://source:PASSWORD at 74.208.297.***:****/stream.mp3"/ >

It's not doing anything do I have to already have stream.mp3 file created on
the icecast server or will it be created by itself  I have the server set
for mountpoint as /stream

I have it setup to record locally when host hits "record" now to get it to
the stream so that people using the website player can hear the conference
call from the icecast server at the same time.

"conference foo-10.10.10.1 record shout://foo.bar.com:8000/foobar.mp3"

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Setup-help-Conference-3-way-call-FIFO-MOH-record-to-Shoutcast-tp6386123p6456754.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From peter.olsson at visionutveckling.se  Thu Jun  9 11:18:55 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 9 Jun 2011 09:18:55 +0200
Subject: [Freeswitch-users] Sofia invite issues
In-Reply-To: <BANLkTikWTD7bT+kiMdDk1VTOU2XsPU0rcQ@mail.gmail.com>
References: <BANLkTikWTD7bT+kiMdDk1VTOU2XsPU0rcQ@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59DEE58301@cooper>

Please pastebin a complete debug log, with "sofia global siptrace on", to http://pastebin.freeswitch.org.

I'm pretty sure this is not a bug in FS, if it was, we would have lots of people complaining...

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Kurtis Heimerl
Skickat: den 8 juni 2011 23:32
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] Sofia invite issues

Hello Freeswitch-users!

I'm currently writing a SIP middlebox that intercepts sip messages and
changes their username. The reasons for this are complicated and
somewhat outside of the scope of this discussion. However, I've come
upon a very strange issue: when making a phone-to-phone call across
freeswitch; it is connecting the wrong user. I'm fairly well convinced
this is a bug, but I thought I'd send the issue here and see if it's
anything I'm obviously doing wrong.

Basically, FS sees a SIP message from a registered UA (call it A)
inviting another registered user to a call (B). This is acked
correctly (a TRYING message). FS then responds by inviting B into a
call with some OTHER user C. C is not mentioned at any point in the
initial SIP messages (as verified by wireshark). Even more strangely,
if I remove user C from FS (by removing their config file) FS responds
to the invite by inviting B into a call with itself. In each case, the
appropriate from header in the invite should be the original caller A.
 It's worth noting that I am not monkeying with the RTP packets at
all, but my understanding is that SIP signalling shouldn't be affected
by that.

My general guess is that I'm messing up the naming somehow, and FS is
running an algorithm to guess at who the call originator is. However,
the naming must be roughly correct; Asterisk is able to handle this
call just fine. I've included a sip trace of the second situation (A
calls B, FS invites B to a call with itself)

For the record A:1300
B:1301
C:IMSI641104878332498

REGISTER MESSAGE (Which works fine)

REGISTER sip:192.168.1.144 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
From: 1300 <sip:1300 at 192.168.1.144>;tag=ftoui
To: 1300 <sip:1300 at 192.168.1.144>
Call-ID: 1032827938 at 192.168.1.144
CSeq: 91 REGISTER
Contact: <sip:1300 at 192.168.1.144:5063>;expires=7200
Max-Forwards: 70
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
From: 1300 <sip:1300 at 192.168.1.144>;tag=ftoui
To: 1300 <sip:1300 at 192.168.1.144>;tag=jU64NXypQc57F
Call-ID: 1032827938 at 192.168.1.144
CSeq: 91 REGISTER
Contact: <sip:1300 at 192.168.1.144:5063>;expires=7200
Date: Wed, 08 Jun 2011 21:02:04 GMT
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-71d64e6 2011-06-07 22-30-16 -0700
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Length: 0

SIP TRACE (A -> B, B -> B response)

INITIAL INVITE
INVITE sip:1301 at 192.168.1.144 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK77974
From: 1300 <sip:1300 at 192.168.1.144>;tag=bgdqx
To: <sip:1301 at 192.168.1.144>
Call-ID: 1817795092 at 192.168.1.144
CSeq: 485 INVITE
Contact: <sip:1300 at 192.168.1.144:5063>;expires=3600
Content-Type: application/sdp
Max-Forwards: 70
Content-Length: 143

INVITE ACK
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK77974
From: 1300 <sip:1300 at 192.168.1.144>;tag=bgdqx
To: <sip:1301 at 192.168.1.144>
Call-ID: 1817795092 at 192.168.1.144
CSeq: 485 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-71d64e6 2011-06-07 22-30-16 -0700
Content-Length: 0

FS INVITE (Note the from address being wrong)
INVITE sip:1301 at 192.168.1.144:5063 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.144;rport;branch=z9hG4bKeeQvcyZ70SDUg
Max-Forwards: 69
From: "1301" <sip:1301 at 192.168.1.144>;tag=rHyS0Z3B61arN
To: <sip:1301 at 192.168.1.144:5063>
Call-ID: 84df47f9-0cb7-122f-13b5-5cff350d9de5
CSeq: 13447852 INVITE
Contact: <sip:mod_sofia at 192.168.1.144:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-71d64e6 2011-06-07 22-30-16 -0700
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
sla, include-session-description, presence.winfo, message-summary,
refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 315
X-FS-Support: update_display
Remote-Party-ID: "1301"
<sip:1301 at 192.168.1.144>;party=calling;screen=yes;privacy=off

USER CONFIGURATIONS

1300.xml
<include>
  <user id="1300" cidr="192.168.1.0/24,127.0.0.1/32">
    <params>
      <param name="accept-blind-reg" value="true"/>
      <param name="accept-blind-auth" value="true"/>
      <param name="vm-password" value="1000"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="1300"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="1300"/>
      <variable name="effective_caller_id_number" value="1300"/>
      <variable name="outbound_caller_id_name" value="1300"/>
      <variable name="outbound_caller_id_number" value="1300"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>

1301.xml
<include>
  <user id="1301" cidr="192.168.1.0/24,127.0.0.1/32">
    <params>
      <param name="accept-blind-reg" value="true"/>
      <param name="accept-blind-auth" value="true"/>
      <param name="vm-password" value="1000"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="1301"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="1301"/>
      <variable name="effective_caller_id_number" value="1301"/>
      <variable name="outbound_caller_id_name" value="1301"/>
      <variable name="outbound_caller_id_number" value="1301"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>

IMSI641104878332498.xml
<include>
  <user id="IMSI641104878332498" cidr="192.168.1.0/24,127.0.0.1/32"
number-alias="1302">
    <params>
      <param name="accept-blind-reg" value="true"/>
      <param name="accept-blind-auth" value="true"/>
      <param name="vm-password" value="1000"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="1302"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="IMSI641104878332498"/>
      <variable name="effective_caller_id_number" value="1302"/>
      <variable name="outbound_caller_id_name" value="IMSI641104878332498"/>
      <variable name="outbound_caller_id_number" value="1302"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>

DIALPLAN

    <extension name="Local Numbers">
      <condition field="destination_number" expression="^130[0-9]$">
	<!--<action application="bridge"
data="user/${user_data(${destination_number}@${domain_name} attr
id)}"/> -->
	<action application="bridge" data="user/${destination_number}"/>
      </condition>
    </extension>

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

!DSPAM:4defeb7432764293448790!



From freeswitch at earthspike.net  Thu Jun  9 11:22:43 2011
From: freeswitch at earthspike.net (John)
Date: Thu, 09 Jun 2011 08:22:43 +0100
Subject: [Freeswitch-users] FreeTDM - Sangoma B700 - ISDN connection
 questions - UK
In-Reply-To: <BANLkTimpvqDfhKEbF=AgaVC3fy1oebY84A@mail.gmail.com>
References: <4DED4854.6010203@earthspike.net>
	<BANLkTimpvqDfhKEbF=AgaVC3fy1oebY84A@mail.gmail.com>
Message-ID: <4DF074C3.9080707@earthspike.net>

Sidharth,

You are right. I'm new to the list and have since found pastebin.

Thank you.

John

On 09/06/11 04:21, Sidharth Kshatriya wrote:
> Dear John,
>
> Appreciate the long time you took in putting down your queries. I'd 
> like to help but unfortunately don't have domain expertise in the 
> areas you're having problems.
>
> Your mail is more likely to be read if it was shorter. May I suggest 
> you use http://pastebin.freeswitch.org/ to paste all your debug 
> messages and configurations.
>
> Please ignore this message if you don't agree with the strategy.
>
> Thanks,
>
> Sidharth
>
> On Tue, Jun 7, 2011 at 3:06 AM, John <freeswitch at earthspike.net 
> <mailto:freeswitch at earthspike.net>> wrote:
>
>     Hello,
>
[ ... ]
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From a.luppi at seletech.com  Thu Jun  9 11:35:53 2011
From: a.luppi at seletech.com (Alessandro)
Date: Thu, 09 Jun 2011 09:35:53 +0200
Subject: [Freeswitch-users] Response status from client
In-Reply-To: <BANLkTim5=tvyxYJgq3AhtfMd2=NHWvQx+Q@mail.gmail.com>
References: <4DEFCA0F.7090808@seletech.com>
	<BANLkTim5=tvyxYJgq3AhtfMd2=NHWvQx+Q@mail.gmail.com>
Message-ID: <4DF077D9.2090208@seletech.com>

Hi,

the url is: http://pastebin.freeswitch.org/16463
i made a call from phone1 to phone2, the called party refused the call 
with code 603. FS received the status 603 form the called (softphone 2) 
party. Than FS sent to the calling party (softphone 1) the message 200 
and bye.

This is the resume of the log:

1000 at localnet_ip                      FS(ip:localnet_ip)               
                        1001 at localnet_ip

INVITE ---------->
                                      INVITE --------------->
<-------------- trying<--------------------trying
                                      <------------------  603

<-------------- 200

ACK------------------>
<--------------------BYE



/You said you had voicemail before... you can't send 603 back to the 
client and continue to voicemail because the 603 terminates the call./

When the called party terminates the call before answering, the calling 
party receive e registered message like "The phone called is not 
available, leave a message ...".  Than i found the registered message in 
freeswitch. (I'm using fusion-pbx)


Thanks

Regards

Alessandro


Il 08/06/2011 21:55, Steven Ayre ha scritto:
>
>     Question 1:
>     i'm developing a custom client sip with pjsip. This client when
>     receive a call that can't be accepted respond with status 603. I
>     think that freeswitch filter this status.
>
>
> 603 gets treated fine for me. I think we need to see more information 
> - can you put a debug level log of the call with siptrace enabled 
> (sofia global siptrace on) on pastebin 
> (http://pastebin.freeswitch.org/) and then post the url here?
>
> Chances are you're doing something in the dialplan that's answering 
> the call, either before or after the failed bleg.
>
> You said you had voicemail before... you can't send 603 back to the 
> client and continue to voicemail because the 603 terminates the call.
>
>     Question:2
>
>     It's possible a custom Header pass trough in status response like
>     trying or session in progress? I'm able to use custom header only
>     on invite adding to invite a header with name like X-myheader. Any
>     suggestion?
>
>
> Yes, you can for 180/183, with the sip_ph_X- prefix. That puts the 
> header on any provisional response.
> http://wiki.freeswitch.org/wiki/Sofia#Adding_Response_Headers
>
> For example:
> <action application="set" data="sip_p_X-myheader=value"/>
>
> AFAIK you won't be able to do the same for a 100 Trying since Sofia 
> doesn't let FS do any handling at the required point. But have a try 
> anyway just to be sure.
>
> -Steve
>
>
> On 8 June 2011 20:14, Alessandro <a.luppi at seletech.com 
> <mailto:a.luppi at seletech.com>> wrote:
>
>     Hi,
>
>     I have two questions about FS:
>
>     Question 1:
>     i'm developing a custom client sip with pjsip. This client when
>     receive a call that can't be accepted respond with status 603. I
>     think that freeswitch filter this status.
>     This is an example of desired behaviour:
>     1000 at localnet_ip                      FS(ip:localnet_ip)          
>                                1001 at localnet_ip
>
>     INVITE ---------->
>                                           INVITE --------------->
>     <-------------- trying<--------------------trying
>     <-------------- 603<------------------  603
>     ACK ------------>                     ACK------------------>
>
>     The current behaviour of FS is:
>
>     1000 at localnet_ip                      FS(ip:localnet_ip)          
>                                1001 at localnet_ip
>
>     INVITE ---------->
>                                           INVITE --------------->
>     <-------------- trying<--------------------trying
>                                           <------------------  603
>
>     <-------------- 200
>
>     ACK------------------>
>     <--------------------BYE
>
>     I'd like to avoid the current behaviour. It's possible a kind of message status path trough?
>     If the called party terminate the call before answering, FS send always to the calling partner 200 and BYE.
>     First thought was related to the voice-mail. Now voice-mail is disabled but the behaviour is the same.
>
>     Question:2
>     It's possible a custom Header pass trough in status response like trying or session in progress?
>     I'm able to use custom header only on invite adding to invite a header with name like X-myheader.
>     Any suggestion?
>
>     Thanks
>     Good Evening
>
>     Alessandro
>
>     -- 
>     Ing. Alessandro Luppi
>     Software development
>     Seletech srl
>     Via Collodi 8, 20052 Monza (MI) - Italy
>     Tel: +39.039.5962000 - Fax: +39.039.9716905
>     email:a.luppi at seletech.com  <mailto:a.luppi at seletech.com>  - Web:www.seletech.com  <http://www.seletech.com>    orwww.seletech.eu  <http://www.seletech.eu>
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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

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From steveayre at gmail.com  Thu Jun  9 12:11:53 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 9 Jun 2011 09:11:53 +0100
Subject: [Freeswitch-users] Response status from client
In-Reply-To: <4DF077D9.2090208@seletech.com>
References: <4DEFCA0F.7090808@seletech.com>
	<BANLkTim5=tvyxYJgq3AhtfMd2=NHWvQx+Q@mail.gmail.com>
	<4DF077D9.2090208@seletech.com>
Message-ID: <BANLkTimELhf+gRJZSbmLvMgzFnOQA6Deww@mail.gmail.com>

I don't see the siptrace in that log?

-Steve


On 9 June 2011 08:35, Alessandro <a.luppi at seletech.com> wrote:

>  Hi,
>
> the url is: http://pastebin.freeswitch.org/16463
> i made a call from phone1 to phone2, the called party refused the call with
> code 603. FS received the status 603 form the called (softphone 2) party.
> Than FS sent to the calling party (softphone 1) the message 200 and bye.
>
> This is the resume of the log:
>
>
> 1000 at localnet_ip                      FS(ip:localnet_ip)
>                        1001 at localnet_ip
>
>  INVITE ---------->
>                                      INVITE --------------->
> <-------------- trying                <--------------------trying
>                                      <------------------  603
>
> <-------------- 200
>
> ACK------------------> <--------------------BYE
>
>
>
> *You said you had voicemail before... you can't send 603 back to the
> client and continue to voicemail because the 603 terminates the call.*
>
> When the called party terminates the call before answering, the calling
> party receive e registered message like "The phone called is not available,
> leave a message ...".  Than i found the registered message in freeswitch.
> (I'm using fusion-pbx)
>
>
> Thanks
>
> Regards
>
> Alessandro
>
>
> Il 08/06/2011 21:55, Steven Ayre ha scritto:
>
>  Question 1:
>> i'm developing a custom client sip with pjsip. This client when receive a
>> call that can't be accepted respond with status 603. I think that freeswitch
>> filter this status.
>>
>
> 603 gets treated fine for me. I think we need to see more information - can
> you put a debug level log of the call with siptrace enabled (sofia global
> siptrace on) on pastebin (http://pastebin.freeswitch.org/) and then post
> the url here?
>
> Chances are you're doing something in the dialplan that's answering the
> call, either before or after the failed bleg.
>
> You said you had voicemail before... you can't send 603 back to the client
> and continue to voicemail because the 603 terminates the call.
>
>
>> Question:2
>>
> It's possible a custom Header pass trough in status response like trying or
>> session in progress? I'm able to use custom header only on invite adding to
>> invite a header with name like X-myheader. Any suggestion?
>>
>
> Yes, you can for 180/183, with the sip_ph_X- prefix. That puts the header
> on any provisional response.
> http://wiki.freeswitch.org/wiki/Sofia#Adding_Response_Headers
>
> For example:
> <action application="set" data="sip_p_X-myheader=value"/>
>
> AFAIK you won't be able to do the same for a 100 Trying since Sofia doesn't
> let FS do any handling at the required point. But have a try anyway just to
> be sure.
>
>  -Steve
>
>
> On 8 June 2011 20:14, Alessandro <a.luppi at seletech.com> wrote:
>
>>  Hi,
>>
>> I have two questions about FS:
>>
>> Question 1:
>> i'm developing a custom client sip with pjsip. This client when receive a
>> call that can't be accepted respond with status 603. I think that freeswitch
>> filter this status.
>> This is an example of desired behaviour:
>>  1000 at localnet_ip                      FS(ip:localnet_ip)
>>                        1001 at localnet_ip
>>
>>  INVITE ---------->
>>                                      INVITE --------------->
>> <-------------- trying                <--------------------trying
>> <-------------- 603                  <------------------  603
>> ACK ------------>                    ACK------------------>
>>
>>
>>
>> The current behaviour of FS is:
>>
>>  1000 at localnet_ip                      FS(ip:localnet_ip)
>>                        1001 at localnet_ip
>>
>>  INVITE ---------->
>>                                      INVITE --------------->
>> <-------------- trying                <--------------------trying
>>                                      <------------------  603
>>
>> <-------------- 200
>>
>> ACK------------------> <--------------------BYE I'd like to avoid the
>> current behaviour. It's possible a kind of message status path trough? If
>> the called party terminate the call before answering, FS send always to the
>> calling partner 200 and BYE. First thought was related to the voice-mail.
>> Now voice-mail is disabled but the behaviour is the same. Question:2 It's
>> possible a custom Header pass trough in status response like trying or
>> session in progress? I'm able to use custom header only on invite adding to
>> invite a header with name like X-myheader. Any suggestion? Thanks Good
>> Evening
>>
>>  Alessandro
>>
>> --
>> Ing. Alessandro Luppi
>> Software development
>> Seletech srl
>> Via Collodi 8, 20052 Monza (MI) - Italy
>> Tel: +39.039.5962000 - Fax: +39.039.9716905
>> email: a.luppi at seletech.com - Web: www.seletech.com   or   www.seletech.eu
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> Ing. Alessandro Luppi
> Software development
> Seletech srl
> Via Collodi 8, 20052 Monza (MI) - Italy
> Tel: +39.039.5962000 - Fax: +39.039.9716905
> email: a.luppi at seletech.com - Web: www.seletech.com   or   www.seletech.eu
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Thu Jun  9 12:16:05 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 9 Jun 2011 09:16:05 +0100
Subject: [Freeswitch-users] Response status from client
In-Reply-To: <BANLkTimELhf+gRJZSbmLvMgzFnOQA6Deww@mail.gmail.com>
References: <4DEFCA0F.7090808@seletech.com>
	<BANLkTim5=tvyxYJgq3AhtfMd2=NHWvQx+Q@mail.gmail.com>
	<4DF077D9.2090208@seletech.com>
	<BANLkTimELhf+gRJZSbmLvMgzFnOQA6Deww@mail.gmail.com>
Message-ID: <BANLkTi=nn25gHEANH2nqadhiN3ycKki2dQ@mail.gmail.com>

It's a mistake in your dialplan. See lines 342-343:

   1. Dialplan: sofia/internal/1009 at 192.168.2.101 Action bridge(user/${
   dialed_extension}@${domain_name})
   2. Dialplan: sofia/internal/1009 at 192.168.2.101 Action answer()
   3. Dialplan: sofia/internal/1009 at 192.168.2.101 Action sleep(1000)
   4. Dialplan: sofia/internal/1009 at 192.168.2.101 Action voicemail(default $
   {domain_name} ${dialed_extension})


You're answering the call after the bridge before it goes to voicemail. That
answer is what is generating the 200.

See lines 507-508 to see it happening:

   1. 2011-06-09 09:15:15.627871 [INFO] mod_dptools.c:2393 Originate
   Failed.  Cause: CALL_REJECTED
   2. EXECUTE sofia/internal/1009 at 192.168.2.101 answer()

-Steve



On 9 June 2011 09:11, Steven Ayre <steveayre at gmail.com> wrote:

> I don't see the siptrace in that log?
>
> -Steve
>
>
> On 9 June 2011 08:35, Alessandro <a.luppi at seletech.com> wrote:
>
>>  Hi,
>>
>> the url is: http://pastebin.freeswitch.org/16463
>> i made a call from phone1 to phone2, the called party refused the call
>> with code 603. FS received the status 603 form the called (softphone 2)
>> party. Than FS sent to the calling party (softphone 1) the message 200 and
>> bye.
>>
>> This is the resume of the log:
>>
>>
>> 1000 at localnet_ip                      FS(ip:localnet_ip)
>>                        1001 at localnet_ip
>>
>>  INVITE ---------->
>>                                      INVITE --------------->
>> <-------------- trying                <--------------------trying
>>                                      <------------------  603
>>
>> <-------------- 200
>>
>> ACK------------------> <--------------------BYE
>>
>>
>>
>> *You said you had voicemail before... you can't send 603 back to the
>> client and continue to voicemail because the 603 terminates the call.*
>>
>> When the called party terminates the call before answering, the calling
>> party receive e registered message like "The phone called is not available,
>> leave a message ...".  Than i found the registered message in freeswitch.
>> (I'm using fusion-pbx)
>>
>>
>> Thanks
>>
>> Regards
>>
>> Alessandro
>>
>>
>> Il 08/06/2011 21:55, Steven Ayre ha scritto:
>>
>>  Question 1:
>>> i'm developing a custom client sip with pjsip. This client when receive a
>>> call that can't be accepted respond with status 603. I think that freeswitch
>>> filter this status.
>>>
>>
>> 603 gets treated fine for me. I think we need to see more information -
>> can you put a debug level log of the call with siptrace enabled (sofia
>> global siptrace on) on pastebin (http://pastebin.freeswitch.org/) and
>> then post the url here?
>>
>> Chances are you're doing something in the dialplan that's answering the
>> call, either before or after the failed bleg.
>>
>> You said you had voicemail before... you can't send 603 back to the client
>> and continue to voicemail because the 603 terminates the call.
>>
>>
>>> Question:2
>>>
>> It's possible a custom Header pass trough in status response like trying
>>> or session in progress? I'm able to use custom header only on invite adding
>>> to invite a header with name like X-myheader. Any suggestion?
>>>
>>
>> Yes, you can for 180/183, with the sip_ph_X- prefix. That puts the header
>> on any provisional response.
>> http://wiki.freeswitch.org/wiki/Sofia#Adding_Response_Headers
>>
>> For example:
>> <action application="set" data="sip_p_X-myheader=value"/>
>>
>> AFAIK you won't be able to do the same for a 100 Trying since Sofia
>> doesn't let FS do any handling at the required point. But have a try anyway
>> just to be sure.
>>
>>  -Steve
>>
>>
>> On 8 June 2011 20:14, Alessandro <a.luppi at seletech.com> wrote:
>>
>>>  Hi,
>>>
>>> I have two questions about FS:
>>>
>>> Question 1:
>>> i'm developing a custom client sip with pjsip. This client when receive a
>>> call that can't be accepted respond with status 603. I think that freeswitch
>>> filter this status.
>>> This is an example of desired behaviour:
>>>  1000 at localnet_ip                      FS(ip:localnet_ip)
>>>                        1001 at localnet_ip
>>>
>>>  INVITE ---------->
>>>                                      INVITE --------------->
>>> <-------------- trying                <--------------------trying
>>> <-------------- 603                  <------------------  603
>>> ACK ------------>                    ACK------------------>
>>>
>>>
>>>
>>> The current behaviour of FS is:
>>>
>>>  1000 at localnet_ip                      FS(ip:localnet_ip)
>>>                        1001 at localnet_ip
>>>
>>>  INVITE ---------->
>>>                                      INVITE --------------->
>>> <-------------- trying                <--------------------trying
>>>                                      <------------------  603
>>>
>>> <-------------- 200
>>>
>>> ACK------------------> <--------------------BYE I'd like to avoid the
>>> current behaviour. It's possible a kind of message status path trough? If
>>> the called party terminate the call before answering, FS send always to the
>>> calling partner 200 and BYE. First thought was related to the voice-mail.
>>> Now voice-mail is disabled but the behaviour is the same. Question:2 It's
>>> possible a custom Header pass trough in status response like trying or
>>> session in progress? I'm able to use custom header only on invite adding to
>>> invite a header with name like X-myheader. Any suggestion? Thanks Good
>>> Evening
>>>
>>>  Alessandro
>>>
>>> --
>>> Ing. Alessandro Luppi
>>> Software development
>>> Seletech srl
>>> Via Collodi 8, 20052 Monza (MI) - Italy
>>> Tel: +39.039.5962000 - Fax: +39.039.9716905
>>> email: a.luppi at seletech.com - Web: www.seletech.com   or   www.seletech.eu
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> --
>> Ing. Alessandro Luppi
>> Software development
>> Seletech srl
>> Via Collodi 8, 20052 Monza (MI) - Italy
>> Tel: +39.039.5962000 - Fax: +39.039.9716905
>> email: a.luppi at seletech.com - Web: www.seletech.com   or   www.seletech.eu
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From mtaylor at employees.org  Thu Jun  9 12:42:58 2011
From: mtaylor at employees.org (Mike Taylor)
Date: Thu, 09 Jun 2011 20:42:58 +1200
Subject: [Freeswitch-users] Conceptual/design question - per
 gateway/DTMF & PSQL
In-Reply-To: <BANLkTikcWbjCHnpefni6cL7ttY5p3On_sw@mail.gmail.com>
References: <CA152D11.2FFDF%mike.taylor@staff.snap.net.nz>	<CA152D95.2FFE8%mike.taylor@staff.snap.net.nz>
	<BANLkTikcWbjCHnpefni6cL7ttY5p3On_sw@mail.gmail.com>
Message-ID: <4DF08792.8090801@employees.org>

Thanks both for the pointers,

I've got close to 60 gateways under my internal profile, so tweaking the 
database is definitely preferable to editing xml files (for me)

I've found some excellent documentation on lcr.conf.xml, custom SQL 
queries, and exporting extra variables.
  I tested, then modified my default mod_lcr query, just to simplify 
things, extended my carrier.gateway schema, and successfully (according 
to debugs, set several dtmf related variables.

So it looks like it might be working in the lab, on Git (about a week 
old), so I'll see what happens with more testing (which options work 
etc). I'm not clear on dtmf and dtmf_relay interaction, so that'll be 
the next thing to nail I think.

Hopefully, once this works, I can offer all sorts of options.

Thanks again, excellent help, your replies let me search for what I needed.

Regards,
Mike

On 8/06/2011 7:59 p.m., Steven Ayre wrote:
> The dtmf_type channel variable can be used to override the profile's 
> dtmf setting. Valid settings are rf2833, info and none. Inband would 
> be more complex, as you'd need to call the 
> start_dtmf_generate application, not sure how you could easily 
> integrate that with mod_lcr. Possibly using the execute_on_x family of 
> channel variables. mod_lcr would need to use custom sql to get the 
> additional dtmf variable.
>
> As David pointed out, mod_lcr already lets you set the codec list for 
> a gateway. late-negotiation isn't *required*, but could be useful. 
> Without it the aleg will pick a codec, then you'll dial out with the 
> codec list to the gateway and transcode if needed. With it, the aleg 
> doesn't pick the codec until the bleg has picked one - which may make 
> transcoding less likely.
>
> You could also setup the gateways as sofia gateways, set the 
> codec/dtmf variables on the gateway and have mod_lcr return the 
> sofia/gateway/gwname/ prefix.
>
> A lot of the behaviour in terms of mod_lcr, execute_on_x etc has 
> changed since 1.0.6 so you might find you have to upgrade to get some 
> of the newer features.
>
> -Steve
>
>
> On 8 June 2011 02:46, Mike Taylor <mtaylor at employees.org 
> <mailto:mtaylor at employees.org>> wrote:
>
>     Hi,
>
>     I hope that I've taught myself enough over the last few weeks that
>     this isn't a blindingly stupid question?
>
>     I am providing services to multiple customers on FS 1.0.6
>
>     I have a basic setup; (which I have inherited, hence the basic
>     questions);
>
>     AS5350--------<external>----- FS-----<internal>------customer GW(s)
>     (No Registration)(No Registration)
>
>     All of the GW info is pulled from a PSWL database. Carriers,
>     carrier_id, lcr table etc.
>
>     The bit I am missing is;
>
>     If all of my GW config is in the PSQL database, and that's all
>     accessed by mod_lcr before routing the call, how do I allow for
>     different DTMF types, codecs etc, PER CUSTOMER GW ?
>
>     Am I stuck with actually defining the gateways statically under
>     /etc/freeswitch/conf/sip_profiles/ (or I see I can doit in the
>     dialplan, but that won't achieve what I need)
>
>     I was hoping to be able to pull the DTMF (and other misc info)
>     from the database, but the more I look into it,  the less likely
>     it is that this will work I think. (mod_lcr is already at the
>     routing stage, can't modify DTMF at that point AFAIK)
>
>     Like I said, hope the question isn't so obvious that it's crazy?
>
>     Regards,
>     Mike
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From kawarod at laposte.net  Thu Jun  9 13:23:57 2011
From: kawarod at laposte.net (kawarod)
Date: Thu, 9 Jun 2011 13:23:57 +0400
Subject: [Freeswitch-users] Problem with Cancel
Message-ID: <9406EB1A-D3A4-416C-AF39-A8555D3D329B@laposte.net>

Hi List,

I have an issue with FS connected to a SIP gateway.
When I receive a call from the SIP Gateway and try to hangup the call before he call is answered, so that a Cancel is emitted by the SIP GW, I have a SIP 481 Call Transaction does not exist.

I understand that FS is unable to match this Cancel with the correct Invite, so that FS can not process the Cancel.

Below is the SIP trace. It seems to me that the issue is because the Invite and the Cancel (generated by the same SIP GW) have different branch in the Via field.
But I have no way to update the SIP GW and I absolutely need it for connection to the core network.

May you please help to check if the issue could be related to the Via branch. It seems that they should be the same based on RFC.
Do you have any idea to bypass this issue (any SIP profile magic trick :p).
I made a "make current" this morning to check if this could help.

U 10.69.208.204:5060 -> 10.69.208.200:5063
INVITE sip:770056 at 10.69.208.200:5063;user=phone SIP/2.0.
Via: SIP/2.0/UDP 10.69.208.204:5060;branch=z9hG4bK-600-8524.
To: <sip:770056 at 10.69.208.200:5063;user=phone>.
From: <sip:265912 at 10.69.208.204:5060;user=phone>;tag=s24df07240-600-9408.
Call-ID: 10.69.208.204:5060-4df07240-600-4680.
Max-Forwards: 70.
CSeq: 1 INVITE.
Content-Length: 679.
Remote-Party-Id: <sip:265912 at 10.69.208.204:5060;user=phone>;privacy=off;screen=yes.
X-Fmc-Signaling-Info: location-pdu=83178679040000;location-pdu=689740000.
Contact: <sip:265912 at 10.69.208.204:5060>.
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,REFER,PRACK,INFO,MESSAGE,SUBSCRIBE,NOTIFY,UPDATE.
Content-Type: application/sdp.


U 10.69.208.204:5060 -> 10.69.208.200:5063
CANCEL sip:770056 at 10.69.208.200:5063 SIP/2.0.
Via: SIP/2.0/UDP 10.69.208.204:5060;branch=z9hG4bK-5800-392.
To: <sip:770056 at 10.69.208.200:5063;user=phone>;tag=6XN6KF2N267cp.
From: <sip:265912 at 10.69.208.204:5060;user=phone>;tag=s24df07240-600-9408.
Call-ID: 10.69.208.204:5060-4df07240-600-4680.
Max-Forwards: 70.
CSeq: 1 CANCEL.
Content-Length: 0.
Reason: Q.850;cause=16;text="Normal call clearing".
.


U 10.69.208.200:5063 -> 10.69.208.204:5060
SIP/2.0 481 Call/Transaction Does Not Exist.
Via: SIP/2.0/UDP 10.69.208.204:5060;branch=z9hG4bK-5800-392.
From: <sip:265912 at 10.69.208.204:5060;user=phone>;tag=s24df07240-600-9408.
To: <sip:770056 at 10.69.208.200:5063;user=phone>;tag=6XN6KF2N267cp.
Call-ID: 10.69.208.204:5060-4df07240-600-4680.
CSeq: 1 CANCEL.
Content-Length: 0.

regards,

rod

From eagle.antonio at gmail.com  Thu Jun  9 13:51:51 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Thu, 9 Jun 2011 09:51:51 +0000
Subject: [Freeswitch-users] UUID_Bridge Fails But Reports +OK
In-Reply-To: <BANLkTi=6=vi+MXR4DMgkzZkhuO7nyN-0cw@mail.gmail.com>
References: <BANLkTikrYLH2ae2o7rMZzBmZr7VsbJ==vA@mail.gmail.com>
	<BANLkTi=CfjrRKLGQeP8DMUbxEiqZ99455w@mail.gmail.com>
	<BANLkTi=FOgi3QYwOHUfeEKUe=SdMuSEgEQ@mail.gmail.com>
	<BANLkTimX6S3rSZF8jALpbk0XF1+sQ0daYg@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DE2B62F9@cooper>
	<BANLkTinFdKhyy2HkRLB8Dohtc5CL0vhbiA@mail.gmail.com>
	<BANLkTi=kG-Y0DfQ43Qtkt8u3D2RPji74Tw@mail.gmail.com>
	<BANLkTi=6=vi+MXR4DMgkzZkhuO7nyN-0cw@mail.gmail.com>
Message-ID: <BANLkTi=NEapCB8Y8_R8TBpW9+A_KQJF5pg@mail.gmail.com>

Ok So here i Come with more news regarding this problem :
Based on current GIT HEAD.

FreeSWITCH Version 1.0.head (git-e2ed8c0 2011-06-08 19-32-18 -0500)

Using valet_park together with ESL INBOUND + G711. I'm almost capable of
obtaining 100% ( above 95%) success.

With uuid_bridge + Park()
Even on non-esl generated call just a plain dialplan with Answer() , Park()
, then uuid_bridge on FS console.

Some calls will connect correctly arround 80% while others will freeze in
the following log line SOFT_EXECUTE in mod_sofia
There is no audio between the two phones.
And even reissuing the uuid_bridge command does nothing.


Besides that i'm a happy DEV :D
If you need some more testing drop me an e-mail :)

A/T


2011/5/30 Antonio Teixeira <eagle.antonio at gmail.com>

> Hello Anton.
>
> Still no i? setting a tottaly different enviroment to test other variables
> like network our Voice Provider and also pressure on FS.
> Will keep you guys updated.
>
>
> Regards
> A/T
>
>
> 2011/5/29 Anton VG <anton.vazir at gmail.com>
>
>> Have you resolved the issue?
>>
>> 2011/5/25 Anton VG <anton.vazir at gmail.com>:
>> > Anthony, I do use the same scheme, and did not experienced the
>> > problem. But I do use park everywhere instead of ValetPark - can it be
>> > the reason?
>> >
>> > 2011/5/25 Peter Olsson <peter.olsson at visionutveckling.se>:
>> >> You need to provide the entire debug trace from this, nor just only
>> after calling uuid_bridge.
>> >>
>> >> /Peter
>> >> ________________________________________
>> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [
>> freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Antonio Teixeira
>> [eagle.antonio at gmail.com]
>> >> Skickat: den 25 maj 2011 10:32
>> >> Till: FreeSWITCH Users Help
>> >> ?mne: Re: [Freeswitch-users] UUID_Bridge Fails But Reports +OK
>> >>
>> >> Good Morning List As Promised.
>> >>
>> >> I Have tested with G711 the problem remains and appears not be affected
>> by the codec change.
>> >>
>> >> Here i have my debug log :
>> >> http://pastebin.freeswitch.org/16369
>> >>
>> >> Just a short explanation :
>> >>
>> >> the call XXXX1010 is Leg A and is currently valet_park()
>> >>
>> >> call XXXX371 is Leg B and is currently on park()
>> >>
>> >> Strangely FS says LEG A is out of order but i can hear MOH and
>> sometimes with this exact software it connects fine.
>> >>
>> >> Hope you can help me out i will keep testing during the day.
>> >>
>> >> Regards
>> >> A/T
>> >>
>> >>
>> >> 2011/5/24 Antonio Teixeira <eagle.antonio at gmail.com<mailto:
>> eagle.antonio at gmail.com>>
>> >> Ok Anthony.
>> >>
>> >> I will provide you with data regarding the use of G711 and full debug
>> logs.
>> >> Will keep an eye on the logs.
>> >>
>> >> Regards
>> >> A/T
>> >>
>> >>
>> >> 2011/5/24 Anthony Minessale <anthony.minessale at gmail.com<mailto:
>> anthony.minessale at gmail.com>>
>> >> The +OK only means the attempt to bridge was successful, if something
>> >> else goes wrong after that, you will not know because it happens
>> >> later.
>> >>
>> >> As suggested, look at the cause of the hangup on the failed bridge.
>> >>
>> >>
>> >> On Tue, May 24, 2011 at 4:33 AM, Antonio Teixeira
>> >> <eagle.antonio at gmail.com<mailto:eagle.antonio at gmail.com>> wrote:
>> >>> Hello List , Good Morning.
>> >>>
>> >>> In my fight to get ESL & Python & Freeswitch all to behave properly i
>> >>> noticed a possible BUG ( need the veterans to confirm).
>> >>>
>> >>> Scenario :
>> >>>
>> >>> Originate & ValetPark () + MOH OR  Just Park() , Both show the same
>> problem.
>> >>> Codec G729
>> >>>
>> >>> I Make another call:
>> >>> originate & park()
>> >>> codec G729
>> >>>
>> >>> Now Python ESL Inbound Or FS Console :
>> >>>
>> >>> uuid_bridge uuid1 uuid2
>> >>>
>> >>> +OK uuid
>> >>>
>> >>> Now one of the two things happen :
>> >>>
>> >>> 1) One the call gets connected hurray :) , Audio Perfect  , etc.
>> >>>
>> >>> 2) The call gets dropped :(
>> >>>
>> >>> In both cases uuid_bridge reports +OK even in the case the call is
>> dropped.
>> >>> Even if i Park() Both Calls using the dialplan XML ( NO ESL with those
>> >>> SYN/ASYNC Problems)  I still get sometimes ( not always) a dropped
>> call
>> >>> could this be related to the use of G729 ?
>> >>>
>> >>> I have lots of available licenses.
>> >>>
>> >>> Regards
>> >>> Ant?nio Teixeira
>> >>>
>> >>>
>> >>>
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org<mailto:
>> FreeSWITCH-users at lists.freeswitch.org>
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>>
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
>> >> ClueCon http://www.cluecon.com/
>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>
>> >> AIM: anthm
>> >> MSN:anthony_minessale at hotmail.com<mailto:
>> MSN%3Aanthony_minessale at hotmail.com>
>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:
>> PAYPAL%3Aanthony.minessale at gmail.com>
>> >> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org<mailto:
>> sip%3A888 at conference.freeswitch.org>
>> >> googletalk:conf+888 at conference.freeswitch.org<mailto:
>> googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> >> pstn:+19193869900
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org<mailto:
>> FreeSWITCH-users at lists.freeswitch.org>
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >> !DSPAM:4ddcbfed32761527616399!
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From a.luppi at seletech.com  Thu Jun  9 14:03:13 2011
From: a.luppi at seletech.com (Alessandro)
Date: Thu, 09 Jun 2011 12:03:13 +0200
Subject: [Freeswitch-users] Response status from client
In-Reply-To: <BANLkTi=nn25gHEANH2nqadhiN3ycKki2dQ@mail.gmail.com>
References: <4DEFCA0F.7090808@seletech.com>	<BANLkTim5=tvyxYJgq3AhtfMd2=NHWvQx+Q@mail.gmail.com>	<4DF077D9.2090208@seletech.com>	<BANLkTimELhf+gRJZSbmLvMgzFnOQA6Deww@mail.gmail.com>
	<BANLkTi=nn25gHEANH2nqadhiN3ycKki2dQ@mail.gmail.com>
Message-ID: <4DF09A61.9070801@seletech.com>

Hi,

This is de default diaplan:
http://pastebin.freeswitch.org/16464

but I think that sofia read public dial plan, ths is the public dial plan:

http://pastebin.freeswitch.org/16465


I don't see the point where the FS read operation reported at lines 342-343.

Thanks

Regards

Alessandro



Il 09/06/2011 10:16, Steven Ayre ha scritto:
> It's a mistake in your dialplan. See lines 342-343:
>
>   1.
>       Dialplan: sofia/internal/1009 at 192.168.2.101 Action
>       bridge(user/${dialed_extension}@${domain_name})
>   2.
>       Dialplan: sofia/internal/1009 at 192.168.2.101 Action answer()
>   3.
>       Dialplan: sofia/internal/1009 at 192.168.2.101 Action sleep(1000)
>   4.
>       Dialplan: sofia/internal/1009 at 192.168.2.101 Action
>       voicemail(default ${domain_name} ${dialed_extension})
>
>
> You're answering the call after the bridge before it goes to 
> voicemail. That answer is what is generating the 200.
>
> See lines 507-508 to see it happening:
>
>   1.
>       2011-06-09 09:15:15.627871 [INFO] mod_dptools.c:2393 Originate
>       Failed.  Cause: CALL_REJECTED
>   2.
>       EXECUTE sofia/internal/1009 at 192.168.2.101 answer()
>
> -Steve
>
>
>
> On 9 June 2011 09:11, Steven Ayre <steveayre at gmail.com 
> <mailto:steveayre at gmail.com>> wrote:
>
>     I don't see the siptrace in that log?
>
>     -Steve
>
>
>     On 9 June 2011 08:35, Alessandro <a.luppi at seletech.com
>     <mailto:a.luppi at seletech.com>> wrote:
>
>         Hi,
>
>         the url is: http://pastebin.freeswitch.org/16463
>         i made a call from phone1 to phone2, the called party refused
>         the call with code 603. FS received the status 603 form the
>         called (softphone 2) party. Than FS sent to the calling party
>         (softphone 1) the message 200 and bye.
>
>         This is the resume of the log:
>
>
>         1000 at localnet_ip                      FS(ip:localnet_ip)      
>                                        1001 at localnet_ip
>
>         INVITE ---------->
>                                               INVITE --------------->
>         <-------------- trying<--------------------trying
>                                               <------------------  603
>
>         <-------------- 200
>
>         ACK------------------>
>         <--------------------BYE
>
>
>
>         /You said you had voicemail before... you can't send 603 back
>         to the client and continue to voicemail because the 603
>         terminates the call./
>
>         When the called party terminates the call before answering,
>         the calling party receive e registered message like "The phone
>         called is not available, leave a message ...".  Than i found
>         the registered message in freeswitch. (I'm using fusion-pbx)
>
>
>         Thanks
>
>         Regards
>
>         Alessandro
>
>
>         Il 08/06/2011 21:55, Steven Ayre ha scritto:
>>
>>             Question 1:
>>             i'm developing a custom client sip with pjsip. This
>>             client when receive a call that can't be accepted respond
>>             with status 603. I think that freeswitch filter this status.
>>
>>
>>         603 gets treated fine for me. I think we need to see more
>>         information - can you put a debug level log of the call with
>>         siptrace enabled (sofia global siptrace on) on pastebin
>>         (http://pastebin.freeswitch.org/) and then post the url here?
>>
>>         Chances are you're doing something in the dialplan that's
>>         answering the call, either before or after the failed bleg.
>>
>>         You said you had voicemail before... you can't send 603 back
>>         to the client and continue to voicemail because the 603
>>         terminates the call.
>>
>>             Question:2
>>
>>             It's possible a custom Header pass trough in status
>>             response like trying or session in progress? I'm able to
>>             use custom header only on invite adding to invite a
>>             header with name like X-myheader. Any suggestion?
>>
>>
>>         Yes, you can for 180/183, with the sip_ph_X- prefix. That
>>         puts the header on any provisional response.
>>         http://wiki.freeswitch.org/wiki/Sofia#Adding_Response_Headers
>>
>>         For example:
>>         <action application="set" data="sip_p_X-myheader=value"/>
>>
>>         AFAIK you won't be able to do the same for a 100 Trying since
>>         Sofia doesn't let FS do any handling at the required point.
>>         But have a try anyway just to be sure.
>>
>>         -Steve
>>
>>
>>         On 8 June 2011 20:14, Alessandro <a.luppi at seletech.com
>>         <mailto:a.luppi at seletech.com>> wrote:
>>
>>             Hi,
>>
>>             I have two questions about FS:
>>
>>             Question 1:
>>             i'm developing a custom client sip with pjsip. This
>>             client when receive a call that can't be accepted respond
>>             with status 603. I think that freeswitch filter this status.
>>             This is an example of desired behaviour:
>>             1000 at localnet_ip                      FS(ip:localnet_ip)
>>                                                  1001 at localnet_ip
>>
>>             INVITE ---------->
>>                                                   INVITE --------------->
>>             <-------------- trying<--------------------trying
>>             <-------------- 603<------------------  603
>>             ACK ------------>                     ACK------------------>
>>
>>             The current behaviour of FS is:
>>
>>             1000 at localnet_ip                      FS(ip:localnet_ip)
>>                                                  1001 at localnet_ip
>>
>>             INVITE ---------->
>>                                                   INVITE --------------->
>>             <-------------- trying<--------------------trying
>>                                                   <------------------  603
>>
>>             <-------------- 200
>>
>>             ACK------------------>
>>             <--------------------BYE
>>
>>             I'd like to avoid the current behaviour. It's possible a kind of message status path trough?
>>             If the called party terminate the call before answering, FS send always to the calling partner 200 and BYE.
>>             First thought was related to the voice-mail. Now voice-mail is disabled but the behaviour is the same.
>>
>>             Question:2
>>             It's possible a custom Header pass trough in status response like trying or session in progress?
>>             I'm able to use custom header only on invite adding to invite a header with name like X-myheader.
>>             Any suggestion?
>>
>>             Thanks
>>             Good Evening
>>
>>             Alessandro
>>
>>             -- 
>>             Ing. Alessandro Luppi
>>             Software development
>>             Seletech srl
>>             Via Collodi 8, 20052 Monza (MI) - Italy
>>             Tel: +39.039.5962000 - Fax: +39.039.9716905
>>             email:a.luppi at seletech.com  <mailto:a.luppi at seletech.com>  - Web:www.seletech.com  <http://www.seletech.com>    orwww.seletech.eu  <http://www.seletech.eu>
>>
>>
>>             _______________________________________________
>>             FreeSWITCH-users mailing list
>>             FreeSWITCH-users at lists.freeswitch.org
>>             <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>             http://www.freeswitch.org
>>
>>
>>
>>         _______________________________________________
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>
>
>         -- 
>         Ing. Alessandro Luppi
>         Software development
>         Seletech srl
>         Via Collodi 8, 20052 Monza (MI) - Italy
>         Tel: +39.039.5962000 - Fax: +39.039.9716905
>         email:a.luppi at seletech.com  <mailto:a.luppi at seletech.com>  - Web:www.seletech.com  <http://www.seletech.com>    orwww.seletech.eu  <http://www.seletech.eu>
>
>
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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

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From a.luppi at seletech.com  Thu Jun  9 14:48:38 2011
From: a.luppi at seletech.com (Alessandro)
Date: Thu, 09 Jun 2011 12:48:38 +0200
Subject: [Freeswitch-users] Response status from client
In-Reply-To: <4DF09A61.9070801@seletech.com>
References: <4DEFCA0F.7090808@seletech.com>	<BANLkTim5=tvyxYJgq3AhtfMd2=NHWvQx+Q@mail.gmail.com>	<4DF077D9.2090208@seletech.com>	<BANLkTimELhf+gRJZSbmLvMgzFnOQA6Deww@mail.gmail.com>	<BANLkTi=nn25gHEANH2nqadhiN3ycKki2dQ@mail.gmail.com>
	<4DF09A61.9070801@seletech.com>
Message-ID: <4DF0A506.9040206@seletech.com>

hi,

i found the problem, was in the default.xml dialplan.

Thanks for help

Regards

Alessandro

Il 09/06/2011 12:03, Alessandro ha scritto:
> Hi,
>
> This is de default diaplan:
> http://pastebin.freeswitch.org/16464
>
> but I think that sofia read public dial plan, ths is the public dial plan:
>
> http://pastebin.freeswitch.org/16465
>
>
> I don't see the point where the FS read operation reported at lines 
> 342-343.
>
> Thanks
>
> Regards
>
> Alessandro
>
>
>
> Il 09/06/2011 10:16, Steven Ayre ha scritto:
>> It's a mistake in your dialplan. See lines 342-343:
>>
>>   1.
>>       Dialplan: sofia/internal/1009 at 192.168.2.101 Action
>>       bridge(user/${dialed_extension}@${domain_name})
>>   2.
>>       Dialplan: sofia/internal/1009 at 192.168.2.101 Action answer()
>>   3.
>>       Dialplan: sofia/internal/1009 at 192.168.2.101 Action sleep(1000)
>>   4.
>>       Dialplan: sofia/internal/1009 at 192.168.2.101 Action
>>       voicemail(default ${domain_name} ${dialed_extension})
>>
>>
>> You're answering the call after the bridge before it goes to 
>> voicemail. That answer is what is generating the 200.
>>
>> See lines 507-508 to see it happening:
>>
>>   1.
>>       2011-06-09 09:15:15.627871 [INFO] mod_dptools.c:2393 Originate
>>       Failed.  Cause: CALL_REJECTED
>>   2.
>>       EXECUTE sofia/internal/1009 at 192.168.2.101 answer()
>>
>> -Steve
>>
>>
>>
>> On 9 June 2011 09:11, Steven Ayre <steveayre at gmail.com 
>> <mailto:steveayre at gmail.com>> wrote:
>>
>>     I don't see the siptrace in that log?
>>
>>     -Steve
>>
>>
>>     On 9 June 2011 08:35, Alessandro <a.luppi at seletech.com
>>     <mailto:a.luppi at seletech.com>> wrote:
>>
>>         Hi,
>>
>>         the url is: http://pastebin.freeswitch.org/16463
>>         i made a call from phone1 to phone2, the called party refused
>>         the call with code 603. FS received the status 603 form the
>>         called (softphone 2) party. Than FS sent to the calling party
>>         (softphone 1) the message 200 and bye.
>>
>>         This is the resume of the log:
>>
>>
>>         1000 at localnet_ip                      FS(ip:localnet_ip)    
>>                                          1001 at localnet_ip
>>
>>         INVITE ---------->
>>                                               INVITE --------------->
>>         <-------------- trying<--------------------trying
>>                                               <------------------  603
>>
>>         <-------------- 200
>>
>>         ACK------------------>
>>         <--------------------BYE
>>
>>
>>
>>         /You said you had voicemail before... you can't send 603 back
>>         to the client and continue to voicemail because the 603
>>         terminates the call./
>>
>>         When the called party terminates the call before answering,
>>         the calling party receive e registered message like "The
>>         phone called is not available, leave a message ...".  Than i
>>         found the registered message in freeswitch. (I'm using
>>         fusion-pbx)
>>
>>
>>         Thanks
>>
>>         Regards
>>
>>         Alessandro
>>
>>
>>         Il 08/06/2011 21:55, Steven Ayre ha scritto:
>>>
>>>             Question 1:
>>>             i'm developing a custom client sip with pjsip. This
>>>             client when receive a call that can't be accepted
>>>             respond with status 603. I think that freeswitch filter
>>>             this status.
>>>
>>>
>>>         603 gets treated fine for me. I think we need to see more
>>>         information - can you put a debug level log of the call with
>>>         siptrace enabled (sofia global siptrace on) on pastebin
>>>         (http://pastebin.freeswitch.org/) and then post the url here?
>>>
>>>         Chances are you're doing something in the dialplan that's
>>>         answering the call, either before or after the failed bleg.
>>>
>>>         You said you had voicemail before... you can't send 603 back
>>>         to the client and continue to voicemail because the 603
>>>         terminates the call.
>>>
>>>             Question:2
>>>
>>>             It's possible a custom Header pass trough in status
>>>             response like trying or session in progress? I'm able to
>>>             use custom header only on invite adding to invite a
>>>             header with name like X-myheader. Any suggestion?
>>>
>>>
>>>         Yes, you can for 180/183, with the sip_ph_X- prefix. That
>>>         puts the header on any provisional response.
>>>         http://wiki.freeswitch.org/wiki/Sofia#Adding_Response_Headers
>>>
>>>         For example:
>>>         <action application="set" data="sip_p_X-myheader=value"/>
>>>
>>>         AFAIK you won't be able to do the same for a 100 Trying
>>>         since Sofia doesn't let FS do any handling at the required
>>>         point. But have a try anyway just to be sure.
>>>
>>>         -Steve
>>>
>>>
>>>         On 8 June 2011 20:14, Alessandro <a.luppi at seletech.com
>>>         <mailto:a.luppi at seletech.com>> wrote:
>>>
>>>             Hi,
>>>
>>>             I have two questions about FS:
>>>
>>>             Question 1:
>>>             i'm developing a custom client sip with pjsip. This
>>>             client when receive a call that can't be accepted
>>>             respond with status 603. I think that freeswitch filter
>>>             this status.
>>>             This is an example of desired behaviour:
>>>             1000 at localnet_ip                      FS(ip:localnet_ip)
>>>                                                  1001 at localnet_ip
>>>
>>>             INVITE ---------->
>>>                                                   INVITE --------------->
>>>             <-------------- trying<--------------------trying
>>>             <-------------- 603<------------------  603
>>>             ACK ------------>                     ACK------------------>
>>>
>>>             The current behaviour of FS is:
>>>
>>>             1000 at localnet_ip                      FS(ip:localnet_ip)
>>>                                                  1001 at localnet_ip
>>>
>>>             INVITE ---------->
>>>                                                   INVITE --------------->
>>>             <-------------- trying<--------------------trying
>>>                                                   <------------------  603
>>>
>>>             <-------------- 200
>>>
>>>             ACK------------------>
>>>             <--------------------BYE
>>>
>>>             I'd like to avoid the current behaviour. It's possible a kind of message status path trough?
>>>             If the called party terminate the call before answering, FS send always to the calling partner 200 and BYE.
>>>             First thought was related to the voice-mail. Now voice-mail is disabled but the behaviour is the same.
>>>
>>>             Question:2
>>>             It's possible a custom Header pass trough in status response like trying or session in progress?
>>>             I'm able to use custom header only on invite adding to invite a header with name like X-myheader.
>>>             Any suggestion?
>>>
>>>             Thanks
>>>             Good Evening
>>>
>>>             Alessandro
>>>
>>>             -- 
>>>             Ing. Alessandro Luppi
>>>             Software development
>>>             Seletech srl
>>>             Via Collodi 8, 20052 Monza (MI) - Italy
>>>             Tel: +39.039.5962000 - Fax: +39.039.9716905
>>>             email:a.luppi at seletech.com  <mailto:a.luppi at seletech.com>  - Web:www.seletech.com  <http://www.seletech.com>    orwww.seletech.eu  <http://www.seletech.eu>
>>>
>>>
>>>             _______________________________________________
>>>             FreeSWITCH-users mailing list
>>>             FreeSWITCH-users at lists.freeswitch.org
>>>             <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>             http://www.freeswitch.org
>>>
>>>
>>>
>>>         _______________________________________________
>>>         FreeSWITCH-users mailing list
>>>         FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>         http://www.freeswitch.org
>>
>>
>>         -- 
>>         Ing. Alessandro Luppi
>>         Software development
>>         Seletech srl
>>         Via Collodi 8, 20052 Monza (MI) - Italy
>>         Tel: +39.039.5962000 - Fax: +39.039.9716905
>>         email:a.luppi at seletech.com  <mailto:a.luppi at seletech.com>  - Web:www.seletech.com  <http://www.seletech.com>    orwww.seletech.eu  <http://www.seletech.eu>
>>
>>
>>         _______________________________________________
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> -- 
> Ing. Alessandro Luppi
> Software development
> Seletech srl
> Via Collodi 8, 20052 Monza (MI) - Italy
> Tel: +39.039.5962000 - Fax: +39.039.9716905
> email:a.luppi at seletech.com  - Web:www.seletech.com    orwww.seletech.eu
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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

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From steveayre at gmail.com  Thu Jun  9 14:52:09 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 9 Jun 2011 11:52:09 +0100
Subject: [Freeswitch-users] Response status from client
In-Reply-To: <4DF09A61.9070801@seletech.com>
References: <4DEFCA0F.7090808@seletech.com>
	<BANLkTim5=tvyxYJgq3AhtfMd2=NHWvQx+Q@mail.gmail.com>
	<4DF077D9.2090208@seletech.com>
	<BANLkTimELhf+gRJZSbmLvMgzFnOQA6Deww@mail.gmail.com>
	<BANLkTi=nn25gHEANH2nqadhiN3ycKki2dQ@mail.gmail.com>
	<4DF09A61.9070801@seletech.com>
Message-ID: <BANLkTikhiubv78xoko5W2Qm=LTrA-Rp-oQ@mail.gmail.com>

It's when it's executing the Local_Extension extension on
http://pastebin.freeswitch.org/16464 lines 741-772:

<extension name="Local_Extension">
          <condition field="destination_number" expression="(^\d{2,7}$)">
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s]<app> -->
        <action application="bind_meta_app" data="1 b s
execute_extension::dx XML features"/>
        <action application="bind_meta_app" data="2 b s record_session::$${
recordings_dir}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${
uuid}.wav"/>
        <action application="bind_meta_app" data="3 b s
execute_extension::cf XML features"/>
        <action application="set" data="ringback=${us-ring}"/>
        <action application="set" data="transfer_ringback=$${hold_music}"/>
        <action application="set" data="call_timeout=30"/>
        <!-- <action application="set" data="sip_exclude_contact=${
network_addr}"/> -->
        <action application="set" data="hangup_after_bridge=true"/>
        <!--<action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/>
-->
        <action application="set" data="continue_on_fail=true"/>
        <action application="hash" data="insert/${domain_name}-call_return/$
{dialed_extension}/${caller_id_number}"/>
        <action application="hash" data="insert/${domain_name}
-last_dial_ext/${dialed_extension}/${uuid}"/>
        <action application="set" data="called_party_callgroup=${user_data($
{dialed_extension}@${domain_name} var callgroup)}"/>
        <!--<action application="export" data="nolocal:sip_secure_media=${
user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>-->
        <action application="hash" data="insert/${domain_name}-last_dial/${
called_party_callgroup}/${uuid}"/>
        <action application="bridge" data="user/${dialed_extension}@${
domain_name}"/>
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <!--<action application="bridge"
data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}
"/>-->
        <action application="voicemail" data="default ${domain_name} ${
dialed_extension}"/>
      </condition>
</extension>

http://pastebin.freeswitch.org/16463 line 327 shows it's this extension

-Steve



On 9 June 2011 11:03, Alessandro <a.luppi at seletech.com> wrote:

>  Hi,
>
> This is de default diaplan:
> http://pastebin.freeswitch.org/16464
>
> but I think that sofia read public dial plan, ths is the public dial plan:
>
> http://pastebin.freeswitch.org/16465
>
>
> I don't see the point where the FS read operation reported at lines
> 342-343.
>
> Thanks
>
> Regards
>
> Alessandro
>
>
>
> Il 09/06/2011 10:16, Steven Ayre ha scritto:
>
> It's a mistake in your dialplan. See lines 342-343:
>
>    1. Dialplan: sofia/internal/1009 at 192.168.2.101 Action bridge(user/${
>    dialed_extension}@${domain_name})
>     2. Dialplan: sofia/internal/1009 at 192.168.2.101 Action answer()
>     3. Dialplan: sofia/internal/1009 at 192.168.2.101 Action sleep(1000)
>     4. Dialplan: sofia/internal/1009 at 192.168.2.101 Action voicemail(default
>    ${domain_name} ${dialed_extension})
>
>
>  You're answering the call after the bridge before it goes to voicemail.
> That answer is what is generating the 200.
>
>  See lines 507-508 to see it happening:
>
>    1. 2011-06-09 09:15:15.627871 [INFO] mod_dptools.c:2393 Originate
>    Failed.  Cause: CALL_REJECTED
>     2. EXECUTE sofia/internal/1009 at 192.168.2.101 answer()
>
>  -Steve
>
>
>
> On 9 June 2011 09:11, Steven Ayre <steveayre at gmail.com> wrote:
>
>> I don't see the siptrace in that log?
>>
>>  -Steve
>>
>>
>> On 9 June 2011 08:35, Alessandro <a.luppi at seletech.com> wrote:
>>
>>>  Hi,
>>>
>>> the url is: http://pastebin.freeswitch.org/16463
>>> i made a call from phone1 to phone2, the called party refused the call
>>> with code 603. FS received the status 603 form the called (softphone 2)
>>> party. Than FS sent to the calling party (softphone 1) the message 200 and
>>> bye.
>>>
>>> This is the resume of the log:
>>>
>>>
>>> 1000 at localnet_ip                      FS(ip:localnet_ip)
>>>                        1001 at localnet_ip
>>>
>>>  INVITE ---------->
>>>                                      INVITE --------------->
>>> <-------------- trying                <--------------------trying
>>>                                      <------------------  603
>>>
>>> <-------------- 200
>>>
>>> ACK------------------> <--------------------BYE
>>>
>>>
>>>
>>>  *You said you had voicemail before... you can't send 603 back to the
>>> client and continue to voicemail because the 603 terminates the call.*
>>>
>>>  When the called party terminates the call before answering, the calling
>>> party receive e registered message like "The phone called is not available,
>>> leave a message ...".  Than i found the registered message in freeswitch.
>>> (I'm using fusion-pbx)
>>>
>>>
>>> Thanks
>>>
>>> Regards
>>>
>>> Alessandro
>>>
>>>
>>> Il 08/06/2011 21:55, Steven Ayre ha scritto:
>>>
>>>  Question 1:
>>>> i'm developing a custom client sip with pjsip. This client when receive
>>>> a call that can't be accepted respond with status 603. I think that
>>>> freeswitch filter this status.
>>>>
>>>
>>> 603 gets treated fine for me. I think we need to see more information -
>>> can you put a debug level log of the call with siptrace enabled (sofia
>>> global siptrace on) on pastebin (http://pastebin.freeswitch.org/) and
>>> then post the url here?
>>>
>>> Chances are you're doing something in the dialplan that's answering the
>>> call, either before or after the failed bleg.
>>>
>>> You said you had voicemail before... you can't send 603 back to the
>>> client and continue to voicemail because the 603 terminates the call.
>>>
>>>
>>>> Question:2
>>>>
>>> It's possible a custom Header pass trough in status response like trying
>>>> or session in progress? I'm able to use custom header only on invite adding
>>>> to invite a header with name like X-myheader. Any suggestion?
>>>>
>>>
>>> Yes, you can for 180/183, with the sip_ph_X- prefix. That puts the header
>>> on any provisional response.
>>> http://wiki.freeswitch.org/wiki/Sofia#Adding_Response_Headers
>>>
>>> For example:
>>> <action application="set" data="sip_p_X-myheader=value"/>
>>>
>>> AFAIK you won't be able to do the same for a 100 Trying since Sofia
>>> doesn't let FS do any handling at the required point. But have a try anyway
>>> just to be sure.
>>>
>>>  -Steve
>>>
>>>
>>> On 8 June 2011 20:14, Alessandro <a.luppi at seletech.com> wrote:
>>>
>>>>  Hi,
>>>>
>>>> I have two questions about FS:
>>>>
>>>> Question 1:
>>>> i'm developing a custom client sip with pjsip. This client when receive
>>>> a call that can't be accepted respond with status 603. I think that
>>>> freeswitch filter this status.
>>>> This is an example of desired behaviour:
>>>>  1000 at localnet_ip                      FS(ip:localnet_ip)
>>>>                        1001 at localnet_ip
>>>>
>>>>  INVITE ---------->
>>>>                                      INVITE --------------->
>>>> <-------------- trying                <--------------------trying
>>>> <-------------- 603                  <------------------  603
>>>> ACK ------------>                    ACK------------------>
>>>>
>>>>
>>>>
>>>> The current behaviour of FS is:
>>>>
>>>>  1000 at localnet_ip                      FS(ip:localnet_ip)
>>>>                        1001 at localnet_ip
>>>>
>>>>  INVITE ---------->
>>>>                                      INVITE --------------->
>>>> <-------------- trying                <--------------------trying
>>>>                                      <------------------  603
>>>>
>>>> <-------------- 200
>>>>
>>>> ACK------------------> <--------------------BYE I'd like to avoid the
>>>> current behaviour. It's possible a kind of message status path trough? If
>>>> the called party terminate the call before answering, FS send always to the
>>>> calling partner 200 and BYE. First thought was related to the voice-mail.
>>>> Now voice-mail is disabled but the behaviour is the same. Question:2 It's
>>>> possible a custom Header pass trough in status response like trying or
>>>> session in progress? I'm able to use custom header only on invite adding to
>>>> invite a header with name like X-myheader. Any suggestion? Thanks Good
>>>> Evening
>>>>
>>>>  Alessandro
>>>>
>>>> --
>>>> Ing. Alessandro Luppi
>>>> Software development
>>>> Seletech srl
>>>> Via Collodi 8, 20052 Monza (MI) - Italy
>>>> Tel: +39.039.5962000 - Fax: +39.039.9716905
>>>> email: a.luppi at seletech.com - Web: www.seletech.com   or   www.seletech.eu
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>
>>>
>>>
>>> --
>>> Ing. Alessandro Luppi
>>> Software development
>>> Seletech srl
>>> Via Collodi 8, 20052 Monza (MI) - Italy
>>> Tel: +39.039.5962000 - Fax: +39.039.9716905
>>> email: a.luppi at seletech.com - Web: www.seletech.com   or   www.seletech.eu
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> Ing. Alessandro Luppi
> Software development
> Seletech srl
> Via Collodi 8, 20052 Monza (MI) - Italy
> Tel: +39.039.5962000 - Fax: +39.039.9716905
> email: a.luppi at seletech.com - Web: www.seletech.com   or   www.seletech.eu
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Thu Jun  9 14:52:59 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 9 Jun 2011 11:52:59 +0100
Subject: [Freeswitch-users] Response status from client
In-Reply-To: <4DF0A506.9040206@seletech.com>
References: <4DEFCA0F.7090808@seletech.com>
	<BANLkTim5=tvyxYJgq3AhtfMd2=NHWvQx+Q@mail.gmail.com>
	<4DF077D9.2090208@seletech.com>
	<BANLkTimELhf+gRJZSbmLvMgzFnOQA6Deww@mail.gmail.com>
	<BANLkTi=nn25gHEANH2nqadhiN3ycKki2dQ@mail.gmail.com>
	<4DF09A61.9070801@seletech.com> <4DF0A506.9040206@seletech.com>
Message-ID: <BANLkTik9=hfQ-g28QEuKti0wAYx3D234_A@mail.gmail.com>

Great. :)

No problem.

-Steve


On 9 June 2011 11:48, Alessandro <a.luppi at seletech.com> wrote:

>  hi,
>
> i found the problem, was in the default.xml dialplan.
>
> Thanks for help
>
> Regards
>
> Alessandro
>
> Il 09/06/2011 12:03, Alessandro ha scritto:
>
> Hi,
>
> This is de default diaplan:
> http://pastebin.freeswitch.org/16464
>
> but I think that sofia read public dial plan, ths is the public dial plan:
>
> http://pastebin.freeswitch.org/16465
>
>
> I don't see the point where the FS read operation reported at lines
> 342-343.
>
> Thanks
>
> Regards
>
> Alessandro
>
>
>
> Il 09/06/2011 10:16, Steven Ayre ha scritto:
>
> It's a mistake in your dialplan. See lines 342-343:
>
>    1. Dialplan: sofia/internal/1009 at 192.168.2.101 Action bridge(user/${
>    dialed_extension}@${domain_name})
>     2. Dialplan: sofia/internal/1009 at 192.168.2.101 Action answer()
>     3. Dialplan: sofia/internal/1009 at 192.168.2.101 Action sleep(1000)
>     4. Dialplan: sofia/internal/1009 at 192.168.2.101 Action voicemail(default
>    ${domain_name} ${dialed_extension})
>
>
>  You're answering the call after the bridge before it goes to voicemail.
> That answer is what is generating the 200.
>
>  See lines 507-508 to see it happening:
>
>    1. 2011-06-09 09:15:15.627871 [INFO] mod_dptools.c:2393 Originate
>    Failed.  Cause: CALL_REJECTED
>     2. EXECUTE sofia/internal/1009 at 192.168.2.101 answer()
>
>  -Steve
>
>
>
> On 9 June 2011 09:11, Steven Ayre <steveayre at gmail.com> wrote:
>
>> I don't see the siptrace in that log?
>>
>>  -Steve
>>
>>
>> On 9 June 2011 08:35, Alessandro <a.luppi at seletech.com> wrote:
>>
>>>  Hi,
>>>
>>> the url is: http://pastebin.freeswitch.org/16463
>>> i made a call from phone1 to phone2, the called party refused the call
>>> with code 603. FS received the status 603 form the called (softphone 2)
>>> party. Than FS sent to the calling party (softphone 1) the message 200 and
>>> bye.
>>>
>>> This is the resume of the log:
>>>
>>>
>>> 1000 at localnet_ip                      FS(ip:localnet_ip)
>>>                        1001 at localnet_ip
>>>
>>>  INVITE ---------->
>>>                                      INVITE --------------->
>>> <-------------- trying                <--------------------trying
>>>                                      <------------------  603
>>>
>>> <-------------- 200
>>>
>>> ACK------------------> <--------------------BYE
>>>
>>>
>>>
>>>  *You said you had voicemail before... you can't send 603 back to the
>>> client and continue to voicemail because the 603 terminates the call.*
>>>
>>>  When the called party terminates the call before answering, the calling
>>> party receive e registered message like "The phone called is not available,
>>> leave a message ...".  Than i found the registered message in freeswitch.
>>> (I'm using fusion-pbx)
>>>
>>>
>>> Thanks
>>>
>>> Regards
>>>
>>> Alessandro
>>>
>>>
>>> Il 08/06/2011 21:55, Steven Ayre ha scritto:
>>>
>>>  Question 1:
>>>> i'm developing a custom client sip with pjsip. This client when receive
>>>> a call that can't be accepted respond with status 603. I think that
>>>> freeswitch filter this status.
>>>>
>>>
>>> 603 gets treated fine for me. I think we need to see more information -
>>> can you put a debug level log of the call with siptrace enabled (sofia
>>> global siptrace on) on pastebin (http://pastebin.freeswitch.org/) and
>>> then post the url here?
>>>
>>> Chances are you're doing something in the dialplan that's answering the
>>> call, either before or after the failed bleg.
>>>
>>> You said you had voicemail before... you can't send 603 back to the
>>> client and continue to voicemail because the 603 terminates the call.
>>>
>>>
>>>> Question:2
>>>>
>>> It's possible a custom Header pass trough in status response like trying
>>>> or session in progress? I'm able to use custom header only on invite adding
>>>> to invite a header with name like X-myheader. Any suggestion?
>>>>
>>>
>>> Yes, you can for 180/183, with the sip_ph_X- prefix. That puts the header
>>> on any provisional response.
>>> http://wiki.freeswitch.org/wiki/Sofia#Adding_Response_Headers
>>>
>>> For example:
>>> <action application="set" data="sip_p_X-myheader=value"/>
>>>
>>> AFAIK you won't be able to do the same for a 100 Trying since Sofia
>>> doesn't let FS do any handling at the required point. But have a try anyway
>>> just to be sure.
>>>
>>>  -Steve
>>>
>>>
>>> On 8 June 2011 20:14, Alessandro <a.luppi at seletech.com> wrote:
>>>
>>>>  Hi,
>>>>
>>>> I have two questions about FS:
>>>>
>>>> Question 1:
>>>> i'm developing a custom client sip with pjsip. This client when receive
>>>> a call that can't be accepted respond with status 603. I think that
>>>> freeswitch filter this status.
>>>> This is an example of desired behaviour:
>>>>  1000 at localnet_ip                      FS(ip:localnet_ip)
>>>>                        1001 at localnet_ip
>>>>
>>>>  INVITE ---------->
>>>>                                      INVITE --------------->
>>>> <-------------- trying                <--------------------trying
>>>> <-------------- 603                  <------------------  603
>>>> ACK ------------>                    ACK------------------>
>>>>
>>>>
>>>>
>>>>
>>>> The current behaviour of FS is:
>>>>
>>>>  1000 at localnet_ip                      FS(ip:localnet_ip)
>>>>                        1001 at localnet_ip
>>>>
>>>>  INVITE ---------->
>>>>                                      INVITE --------------->
>>>> <-------------- trying                <--------------------trying
>>>>                                      <------------------  603
>>>>
>>>> <-------------- 200
>>>>
>>>> ACK------------------> <--------------------BYE I'd like to avoid the
>>>> current behaviour. It's possible a kind of message status path trough? If
>>>> the called party terminate the call before answering, FS send always to the
>>>> calling partner 200 and BYE. First thought was related to the voice-mail.
>>>> Now voice-mail is disabled but the behaviour is the same. Question:2 It's
>>>> possible a custom Header pass trough in status response like trying or
>>>> session in progress? I'm able to use custom header only on invite adding to
>>>> invite a header with name like X-myheader. Any suggestion? Thanks Good
>>>> Evening
>>>>
>>>>  Alessandro
>>>>
>>>> --
>>>> Ing. Alessandro Luppi
>>>> Software development
>>>> Seletech srl
>>>> Via Collodi 8, 20052 Monza (MI) - Italy
>>>> Tel: +39.039.5962000 - Fax: +39.039.9716905
>>>> email: a.luppi at seletech.com - Web: www.seletech.com   or   www.seletech.eu
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>
>>>
>>>
>>> --
>>> Ing. Alessandro Luppi
>>> Software development
>>> Seletech srl
>>> Via Collodi 8, 20052 Monza (MI) - Italy
>>> Tel: +39.039.5962000 - Fax: +39.039.9716905
>>> email: a.luppi at seletech.com - Web: www.seletech.com   or   www.seletech.eu
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> Ing. Alessandro Luppi
> Software development
> Seletech srl
> Via Collodi 8, 20052 Monza (MI) - Italy
> Tel: +39.039.5962000 - Fax: +39.039.9716905
> email: a.luppi at seletech.com - Web: www.seletech.com   or   www.seletech.eu
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> Ing. Alessandro Luppi
> Software development
> Seletech srl
> Via Collodi 8, 20052 Monza (MI) - Italy
> Tel: +39.039.5962000 - Fax: +39.039.9716905
> email: a.luppi at seletech.com - Web: www.seletech.com   or   www.seletech.eu
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Nabble at slickdeals.endjunk.com  Thu Jun  9 15:30:02 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 9 Jun 2011 04:30:02 -0700 (PDT)
Subject: [Freeswitch-users] IP Whitelist
In-Reply-To: <BANLkTi=Monp1fMy0E3Bbq9P5oc-_0X7YFA@mail.gmail.com>
References: <D4A666272A214C4DBBABFE569CA34CA401DFFC7617@doctorbob.loop.local>
	<BANLkTi=Monp1fMy0E3Bbq9P5oc-_0X7YFA@mail.gmail.com>
Message-ID: <1307619002073-6457393.post@n2.nabble.com>


Steven Ayre wrote:
> 
> ACLs control registrations and calls, not options requests.
> 
> You'd be best off blocking sipvicious with this iptables entry:
> 
> iptables -I INPUT -j DROP -p udp --dport 5060 -m string --string
> "friendly-scanner" --algo bm
Will the above IPTABLES work with old versions of sipvicious scan?

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/IP-Whitelist-tp6455077p6457393.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Thu Jun  9 15:37:10 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 9 Jun 2011 12:37:10 +0100
Subject: [Freeswitch-users] IP Whitelist
In-Reply-To: <1307619002073-6457393.post@n2.nabble.com>
References: <D4A666272A214C4DBBABFE569CA34CA401DFFC7617@doctorbob.loop.local>
	<BANLkTi=Monp1fMy0E3Bbq9P5oc-_0X7YFA@mail.gmail.com>
	<1307619002073-6457393.post@n2.nabble.com>
Message-ID: <BANLkTim6j39OKz-WCEdR_Uds+NAr=vm3ZA@mail.gmail.com>

As far as I know all (official) versions of sipvicious use the
friendly-scanner useragent, so will be spotted.

-Steve


On 9 June 2011 12:30, mazilo <Nabble at slickdeals.endjunk.com> wrote:

>
> Steven Ayre wrote:
> >
> > ACLs control registrations and calls, not options requests.
> >
> > You'd be best off blocking sipvicious with this iptables entry:
> >
> > iptables -I INPUT -j DROP -p udp --dport 5060 -m string --string
> > "friendly-scanner" --algo bm
> Will the above IPTABLES work with old versions of sipvicious scan?
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3
> Watts of electricity.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/IP-Whitelist-tp6455077p6457393.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From michal.bielicki at seventhsignal.de  Thu Jun  9 16:49:50 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Thu, 9 Jun 2011 14:49:50 +0200
Subject: [Freeswitch-users] test
Message-ID: <F1DBFF7E-64C9-4B5F-B9C4-BD70D58D5228@seventhsignal.de>

test
Michal Bielicki
Gesch?ftsf?hrer / CEO

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

Amtsgericht Charlottenburg HRA 44413 B
Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de

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From benkokakao at gmail.com  Thu Jun  9 17:50:56 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Thu, 9 Jun 2011 15:50:56 +0200
Subject: [Freeswitch-users] FreeTDM - destination_number empty on
	inbound call
In-Reply-To: <201106082043.54622.stkn@freeswitch.org>
References: <BANLkTimCRbEutxEf2vqt6OrbyVwn0XhHdw@mail.gmail.com>
	<201106082043.54622.stkn@freeswitch.org>
Message-ID: <BANLkTimxNv6=-23CgAfZhMK=nu+G4F89VA@mail.gmail.com>

On 8 June 2011 20:43, Stefan Knoblich <stkn at freeswitch.org> wrote:
> Am Wednesday 08 June 2011 schrieb Christian Benke:
>> My FS is connected to a plain BRI-line via a HFC-S PCI-card. On
>> inbound calls i don't receive a dialed number. Please see the debug at
>> http://pastebin.freeswitch.org/16456 for details.
>> Do i have to fetch the DID from a variable before i can handle the
>> call or is there some other problem, which i assume?
>
> That SETUP message at the top isn't valid, AFAIK.
> There are two 'Calling Number' IEs (0x6c) and the second
> one should have been a 'Called Number' IE (0x70) instead
> ('190341' is the number you dialed, right?).

No, that's actually fine - the first number is user-provided("user
number not screened") as it is a "location independent corporate
number", the second one is the "network provided" regional
number(Which i missed to obfuscate).

After talking with a colleague who is very proficient with ISDN and
testing the line with a ISDN testing device today, we came to the
conclusion that ftdm sends "CALL PROCEEDING" immediately, instead of
sending an "SETUP ACK" and waiting for additional digits. What it
comes down to, i simply didn't allow overlap-dialing.

This was easy to fix for the Sangoma card by setting the
"overlap"-parameter to "yes", as documented in
http://wiki.freeswitch.org/wiki/FreeTDM#FreeSWITCH_FreeTDM_configuration.

However, on my second device, which has a HFC-S based card(OpenVOX
B100P) installed, ftmod_libpri based, overlap-dialing doesn't seem to
work. The debug-logs posted above are from this card. Googling for
overlap-dialing with libpri brought up this thread from Dec 2010,
where Michael Collins says it's not supported yet:
http://lists.freeswitch.org/pipermail/freeswitch-users/2010-December/066695.html
I assume this status has not changed?

Best regards,
Christian


From benkokakao at gmail.com  Thu Jun  9 18:01:31 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Thu, 9 Jun 2011 16:01:31 +0200
Subject: [Freeswitch-users] Proxy traffic security
In-Reply-To: <4DEFF3FA.5050500@gmail.com>
References: <4DEFF3FA.5050500@gmail.com>
Message-ID: <BANLkTincK9zEprZjuMRMo5c6vejuX8_22w@mail.gmail.com>

On 9 June 2011 00:13, Antonio <potxoka at gmail.com> wrote:
> Hello
>
> I have a FreeSwitch configured as a gateway, the proxy makes the user
> authentication and other functions. All servers have public ip's, and I
> have doubts to security. I had thought to put the proxy ip's in acl. Is
> it safe this scenario? Can it be improved? Should we also put the sip
> providers? Thanks.

ACLs definitely would not hurt. Also think about setting up a simple
firewall(e.g. Shorewall or just plain IPTables) to limit the access to
the ip's which MUST have access(e.g. SIP-Provider IPs), you can never
be sure if a software(like FreeSWITCH) doesn't have a bug that could
be exploited.

Cheers,
Christian


From benkokakao at gmail.com  Thu Jun  9 18:03:06 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Thu, 9 Jun 2011 16:03:06 +0200
Subject: [Freeswitch-users] Enumerate variables in a session
In-Reply-To: <33095823FD21DF429B481B5163264B7950AC586589@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950AC586589@VMBX102.ihostexchange.net>
Message-ID: <BANLkTikmrXscsfdPUHhsgewYeF3RYNcyRQ@mail.gmail.com>

On 8 June 2011 20:18, Yungwei Chen <yungwei at resolvity.com> wrote:
> Hi,
>
> I'm wondering if there's way to enumerate the list of variable names in a session in javascript. Thanks.

This is not a FreeSWITCH specific question - a javascript-focused
community or a more general programming website like stackoverflow.com
would be suited better.

Best regards,
Christian


From steveayre at gmail.com  Thu Jun  9 18:17:04 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 9 Jun 2011 15:17:04 +0100
Subject: [Freeswitch-users] Proxy traffic security
In-Reply-To: <4DEFF3FA.5050500@gmail.com>
References: <4DEFF3FA.5050500@gmail.com>
Message-ID: <BANLkTimsyiaWov8GKL8u+dxmqKPOhHFckw@mail.gmail.com>

You can create an ACL listing proxy IPs then set the proxy-acl parameter on
the SIP profile.call

If your proxy can add a X-AUTH-IP header to the INVITE containing the
caller's IP then FS will use that IP to check against ACLs instead of the IP
received from, if the IP received from is on the proxy ACL. That way you can
still authenticate callers with ACLs behind a proxy, although you're
trusting your proxy to set that header correctly. The proxy-acl setting
means only your proxy can set the X-AUTH-IP header, it'll be ignored on
calls from any other IP.

-Steve


On 8 June 2011 23:13, Antonio <potxoka at gmail.com> wrote:

> Hello
>
> I have a FreeSwitch configured as a gateway, the proxy makes the user
> authentication and other functions. All servers have public ip's, and I
> have doubts to security. I had thought to put the proxy ip's in acl. Is
> it safe this scenario? Can it be improved? Should we also put the sip
> providers? Thanks.
>
> Greetings
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Thu Jun  9 18:20:03 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 9 Jun 2011 15:20:03 +0100
Subject: [Freeswitch-users] Enumerate variables in a session
In-Reply-To: <BANLkTikmrXscsfdPUHhsgewYeF3RYNcyRQ@mail.gmail.com>
References: <33095823FD21DF429B481B5163264B7950AC586589@VMBX102.ihostexchange.net>
	<BANLkTikmrXscsfdPUHhsgewYeF3RYNcyRQ@mail.gmail.com>
Message-ID: <BANLkTikMT8GROmAdOnv8vBhuEEBYmiAeYw@mail.gmail.com>

Actually Christian, it is since you have a session object in javascript and
have to access the variables via the Session api.

The simple answer is not directly AFAIK, although I think there is an
undocumented Session.getXMLCDR() function. The XML CDR contains all
variables on the session, so if you parse that XML I think you'll get the
full list.

-Steve


On 9 June 2011 15:03, Christian Benke <benkokakao at gmail.com> wrote:

> On 8 June 2011 20:18, Yungwei Chen <yungwei at resolvity.com> wrote:
> > Hi,
> >
> > I'm wondering if there's way to enumerate the list of variable names in a
> session in javascript. Thanks.
>
> This is not a FreeSWITCH specific question - a javascript-focused
> community or a more general programming website like stackoverflow.com
> would be suited better.
>
> Best regards,
> Christian
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From benkokakao at gmail.com  Thu Jun  9 18:24:38 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Thu, 9 Jun 2011 16:24:38 +0200
Subject: [Freeswitch-users] FS response
In-Reply-To: <BANLkTi=y40L2O4rMcfEonPWte1GjgjnWBQ@mail.gmail.com>
References: <BANLkTi=y40L2O4rMcfEonPWte1GjgjnWBQ@mail.gmail.com>
Message-ID: <BANLkTikgSzkiS6apAd6B-_KtSGvkCiy5uA@mail.gmail.com>

Hmm, i don't believe FS really behaves like the transaction you posted.

Can you provide us with an "original" trace? (Either "sofia global
siptrace on" in fs_cli or install ngrep and do "ngrep -d ethX -W
byline port 5060" on the shell of your freeswitch-host.

Regards
Christian


From benkokakao at gmail.com  Thu Jun  9 18:33:20 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Thu, 9 Jun 2011 16:33:20 +0200
Subject: [Freeswitch-users] Voicemail Message - How t know how many are
 read & how many are unread in Inbox
In-Reply-To: <AA1BE0FA88F3421784DBD1D91A84E515@sharad>
References: <mailman.10243.1307524956.24555.freeswitch-users@lists.freeswitch.org>
	<AA1BE0FA88F3421784DBD1D91A84E515@sharad>
Message-ID: <BANLkTikshbtzM67daHBePp7TB5pH5FjEyA@mail.gmail.com>

> Just wondering whether there is any way to know how many voice messages are
> read or unread in New & SAVED box.

Not sure in "which" way you want to know this. From fs_cli you can try:

> vm_list user at domain xml

This gives you an xml-output of all messages of this user - saved/read
mails have an read_epoch-value, unread messages have a
read_epoch-value of 0, e.g.:

freeswitch at internal> vm_list 200 at 192.168.0.4 xml
<voicemail>
 <message>
  <created_epoch>1307630009</created_epoch>
  <read_epoch>0</read_epoch>
  <username>200</username>
  <domain>192.168.0.4</domain>
  <folder>inbox</folder>
  <path>/usr/local/freeswitch/storage/voicemail/default/192.168.0.4/200/msg_60c27a70-92a5-11e0-8b2f-fd7505ade349.wav</path>
  <uuid>60c27a70-92a5-11e0-8b2f-fd7505ade349</uuid>
  <cid-name>Jim Doe</cid-name>
  <cid-number>300</cid-number>
 </message>
 <message>
  <created_epoch>1307348316</created_epoch>
  <read_epoch>1307629941</read_epoch>
  <username>200</username>
  <domain>192.168.0.4</domain>
  <folder>inbox</folder>
  <path>/usr/local/freeswitch/storage/voicemail/default/192.168.0.4/200/msg_775fa510-9015-11e0-88e7-fd7505ade349.wav</path>
  <uuid>775fa510-9015-11e0-88e7-fd7505ade349</uuid>
  <cid-name>John Doe</cid-name>
  <cid-number>100</cid-number>
 </message>
</voicemail>

freeswitch at internal>

hth
Christian


From steveayre at gmail.com  Thu Jun  9 18:50:14 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 9 Jun 2011 15:50:14 +0100
Subject: [Freeswitch-users] FS response
In-Reply-To: <BANLkTikgSzkiS6apAd6B-_KtSGvkCiy5uA@mail.gmail.com>
References: <BANLkTi=y40L2O4rMcfEonPWte1GjgjnWBQ@mail.gmail.com>
	<BANLkTikgSzkiS6apAd6B-_KtSGvkCiy5uA@mail.gmail.com>
Message-ID: <BANLkTikndJsj8weD_H9yT-rJtA7MRLLTYQ@mail.gmail.com>

He asked a similar question about a 603 that was returning 200 in another
thread, which has already been solved.

I expect it's the same problem, that his dialplan was doing an answer after
the bridge before going to voicemail.

-Steve


On 9 June 2011 15:24, Christian Benke <benkokakao at gmail.com> wrote:

> Hmm, i don't believe FS really behaves like the transaction you posted.
>
> Can you provide us with an "original" trace? (Either "sofia global
> siptrace on" in fs_cli or install ngrep and do "ngrep -d ethX -W
> byline port 5060" on the shell of your freeswitch-host.
>
> Regards
> Christian
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From moises.silva at gmail.com  Thu Jun  9 19:19:03 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 9 Jun 2011 11:19:03 -0400
Subject: [Freeswitch-users] FreeTDM does not work
In-Reply-To: <BANLkTimQnSg5E9aKYeVvfcxxkeiQGvRRHw@mail.gmail.com>
References: <BANLkTimQnSg5E9aKYeVvfcxxkeiQGvRRHw@mail.gmail.com>
Message-ID: <BANLkTikimz4oYs4zPowXMpb0n_SbR5m23g@mail.gmail.com>

2011/6/8 Valery Kalinin <valery.kalinin at gmail.com>

> Hi all!
>
> FreeTDM does not work with Digium equipment.
>
>
That's not true. You just did not configure things correctly.

2011-06-09 02:07:01.693551 [DEBUG] ftmod_libpri.c:1586 -- Caught Event span
> 1 4 (CONFIG_ERR)
> 2011-06-09 02:07:02.773590 [DEBUG] ftmod_libpri.c:1586 -- Caught Event span
> 1 4 (CONFIG_ERR)
> 2011-06-09 02:07:03.833629 [DEBUG] ftmod_libpri.c:1586 -- Caught Event span
> 1 4 (CONFIG_ERR)
> 2011-06-09 02:07:04.893667 [DEBUG] ftmod_libpri.c:146 TEI=0 DL event:
> Q931_DL_EVENT_DL_RELEASE_IND(3)
> 2011-06-09 02:07:05.962709 [DEBUG] ftmod_libpri.c:1586 -- Caught Event span
> 1 4 (CONFIG_ERR)
> 2011-06-09 02:07:07.982780 [DEBUG] ftmod_libpri.c:1586 -- Caught Event span
> 1 4 (CONFIG_ERR)
> 2011-06-09 02:07:10.002854 [DEBUG] ftmod_libpri.c:1586 -- Caught Event span
> 1 4 (CONFIG_ERR)
> ...
>
> Why?
> Which means error: Caught Event span 1 4 (CONFIG_ERR)
> Which config???
>
>
That means libpri does not like the configuration. I just updated the code
to print the error string coming from lbipri. Update to latest GIT and you
will see a more meaningful error. My guess is you configured for network
when you are really the cpe or viceversa.

Moises Silva
Senior Software Engineer, Software Development Manager
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6
Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com
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From a.luppi at seletech.com  Thu Jun  9 19:33:31 2011
From: a.luppi at seletech.com (Alessandro)
Date: Thu, 09 Jun 2011 17:33:31 +0200
Subject: [Freeswitch-users] FS response
In-Reply-To: <BANLkTikndJsj8weD_H9yT-rJtA7MRLLTYQ@mail.gmail.com>
References: <BANLkTi=y40L2O4rMcfEonPWte1GjgjnWBQ@mail.gmail.com>	<BANLkTikgSzkiS6apAd6B-_KtSGvkCiy5uA@mail.gmail.com>
	<BANLkTikndJsj8weD_H9yT-rJtA7MRLLTYQ@mail.gmail.com>
Message-ID: <4DF0E7CB.5010005@seletech.com>

Hi,

yes problem already solved, sorry for the double mail... my mistake...


regards

Alessandro

Il 09/06/2011 16:50, Steven Ayre ha scritto:
> He asked a similar question about a 603 that was returning 200 in 
> another thread, which has already been solved.
>
> I expect it's the same problem, that his dialplan was doing an answer 
> after the bridge before going to voicemail.
>
> -Steve
>
>
> On 9 June 2011 15:24, Christian Benke <benkokakao at gmail.com 
> <mailto:benkokakao at gmail.com>> wrote:
>
>     Hmm, i don't believe FS really behaves like the transaction you
>     posted.
>
>     Can you provide us with an "original" trace? (Either "sofia global
>     siptrace on" in fs_cli or install ngrep and do "ngrep -d ethX -W
>     byline port 5060" on the shell of your freeswitch-host.
>
>     Regards
>     Christian
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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

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From Hector.Geraldino at ip-soft.net  Thu Jun  9 02:14:30 2011
From: Hector.Geraldino at ip-soft.net (Hector Geraldino)
Date: Wed, 8 Jun 2011 18:14:30 -0400
Subject: [Freeswitch-users] originating a call from an outbound socket app
Message-ID: <6A6B4C284AD15042B429EB9D904544AD021F45EAA0@NY1-EXMB-01.ip-soft.net>

Hi all,

I've been playing with FS around weeks, with some level of success. Right now I'm stuck trying to dial an extension, or to bridge a call to an external extension. Briefly, the scenario is the following:


-          Cisco CCM forwarding SIP calls to FS

-          FS in outbound mode,  forwarding calls to a Java application using freeswitch-esl jar

The java application answers the call, says a dialogue using TTS and receives input using DTFM or speech commands; all this is working pretty well. Now I'm trying to put this call in hold, originate a call to another extension and, depending on the result (call answered, not answered, destination user presses an specific number generating an expected DTMF), I need to bridge both calls and finish the session.

I try several ways to get this done with no luck yet: using api originate to make the call with different arguments ({ignore_early_media,  etc). I even tried to skip this step and just do the bridge between the inbound call and the external extension, but this  didn't work either. I can't see any errors thrown by FS, and the result of the command is +OK, but nothing happens, the external phone never rings. The FS log only says:

[DEBUG] switch_core_session.c:954 Send signal sofia/external/5655 at 192.168.8.1 [BREAK] <= this ext/IP address is from the inbound caller, not the one I'm trying to connect to.

My first impression was that I might have something wrong with the originate/bridge parameters. I changed the dialplan XML file to just do the bridge, and it worked like a charm.

<include>
<extension name="test">
        <condition field="destination_number" expression="6114|5121">
<!--    <action application="socket" data="localhost:8044 async full"/>  -->
          <action application="bridge" data="sofia/external/5562 at 192.168.2.14" />
        </condition>
</extension>
</include>

I don't know what path to follow here. I was unable to find relevant examples on google that guides me to solve this little issue. Any advices?

Thanks all,
Hector
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From freeswitch at mralston.com  Thu Jun  9 02:44:43 2011
From: freeswitch at mralston.com (Matthew Ralston)
Date: Wed, 8 Jun 2011 23:44:43 +0100
Subject: [Freeswitch-users] Logical OR in conditions
Message-ID: <43439B44-F6EE-448F-9CB4-58D15ACAE9F6@mralston.com>

Hi,

What is the correct condition->break value to use in order to create a logical OR with conditions?

I have two extensions which I'd like to merge into one. I'd like the actions to be run if either of the conditions matches:

		<extension name="call_forward_enable">
			<condition field="destination_number" expression="^\*21\*([0-9]+)#$">
				<action application="javascript" data="callForwardEnable.js"/>
				<action application="hangup"/>
			</condition>
		</extension>

		<!-- For phones that won't dial *# numbers, *=7, #=9 -->
		<extension name="call_forward_enable">
			<condition field="destination_number" expression="^7217([0-9]+)9$">
				<action application="javascript" data="callForwardEnable.js"/>
				<action application="hangup"/>
			</condition>
		</extension>

So the result would be something like:

		<extension name="call_forward_enable">
			<condition field="destination_number" expression="^\*21\*([0-9]+)#$" break="on-true" />
			<condition field="destination_number" expression="^7217([0-9]+)9$">
				<action application="javascript" data="callForwardEnable.js"/>
				<action application="hangup"/>
			</condition>
		</extension>

I can't seem to get it to work and my brain is melting trying to wrap my head around the different break values.

Cheers,

Matt



From Hector.Geraldino at ip-soft.net  Thu Jun  9 17:12:49 2011
From: Hector.Geraldino at ip-soft.net (Hector Geraldino)
Date: Thu, 9 Jun 2011 09:12:49 -0400
Subject: [Freeswitch-users] originating a call from an outbound socket app
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD021F45EA9E@NY1-EXMB-01.ip-soft.net>
References: <6A6B4C284AD15042B429EB9D904544AD021F45EA9E@NY1-EXMB-01.ip-soft.net>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD021F45EACF@NY1-EXMB-01.ip-soft.net>

Hi all,

I've been playing with FS around weeks, with some level of success. Right now I'm stuck trying to dial an extension, or to bridge a call to an external extension. Briefly, the scenario is the following:


-          Cisco CCM forwarding SIP calls to FS

-          FS in outbound mode,  forwarding calls to a Java application using freeswitch-esl jar

The java application answers the call, says a dialogue using TTS and receives input using DTFM or speech commands; all this is working pretty well. Now I'm trying to put this call in hold, originate a call to another extension and, depending on the result (call answered, not answered, destination user presses an specific number generating an expected DTMF), I need to bridge both calls and finish the session.

I try several ways to get this done with no luck yet: using api originate to make the call with different arguments ({ignore_early_media,  etc). I even tried to skip this step and just do the bridge between the inbound call and the external extension, but this  didn't work either. I can't see any errors thrown by FS, and the result of the command is +OK, but nothing happens, the external phone never rings. The FS log only says:

[DEBUG] switch_core_session.c:954 Send signal sofia/external/5655 at 192.168.8.1 [BREAK] <= this ext/IP address is from the inbound caller, not the one I'm trying to connect to.

My first impression was that I might have something wrong with the originate/bridge parameters. I changed the dialplan XML file to just do the bridge, and it worked like a charm.

<include>
<extension name="test">
        <condition field="destination_number" expression="6114|5121">
<!--    <action application="socket" data="localhost:8044 async full"/>  -->
          <action application="bridge" data="sofia/external/5562 at 192.168.2.14" />
        </condition>
</extension>
</include>

I don't know what path to follow here. I was unable to find relevant examples on google that guides me to solve this little issue. Can somebody please give me some advice?

Thanks all,
Hector
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From freeswitch at mralston.com  Thu Jun  9 19:20:44 2011
From: freeswitch at mralston.com (Matthew Ralston)
Date: Thu, 9 Jun 2011 16:20:44 +0100
Subject: [Freeswitch-users] Logical OR in conditions
Message-ID: <19EE2626-06B6-4971-BFCA-0370EF77A8AC@mralston.com>

Hi,

What is the correct condition->break value to use in order to create a logical OR with conditions?

I have two extensions which I'd like to merge into one. I'd like the actions to be run if either of the conditions matches:

		<extension name="call_forward_enable">
			<condition field="destination_number" expression="^\*21\*([0-9]+)#$">
				<action application="javascript" data="callForwardEnable.js"/>
				<action application="hangup"/>
			</condition>
		</extension>

		<!-- For phones that won't dial *# numbers, *=7, #=9 -->
		<extension name="call_forward_enable">
			<condition field="destination_number" expression="^7217([0-9]+)9$">
				<action application="javascript" data="callForwardEnable.js"/>
				<action application="hangup"/>
			</condition>
		</extension>

So the result would be something like:

		<extension name="call_forward_enable">
			<condition field="destination_number" expression="^\*21\*([0-9]+)#$" break="on-true" />
			<condition field="destination_number" expression="^7217([0-9]+)9$">
				<action application="javascript" data="callForwardEnable.js"/>
				<action application="hangup"/>
			</condition>
		</extension>

I can't seem to get it to work and my brain is melting trying to wrap my head around the different break values.

Cheers,

Matt



From shouldbeq931 at gmail.com  Thu Jun  9 11:27:13 2011
From: shouldbeq931 at gmail.com (shouldbe q931)
Date: Thu, 9 Jun 2011 08:27:13 +0100
Subject: [Freeswitch-users] FreeTDM - Sangoma B700 - ISDN connection
 questions - UK
In-Reply-To: <4DED4854.6010203@earthspike.net>
References: <4DED4854.6010203@earthspike.net>
Message-ID: <BANLkTimP23FVs8Vcg+pbxnfBucrQoJur2w@mail.gmail.com>

On Mon, Jun 6, 2011 at 10:36 PM, John <freeswitch at earthspike.net> wrote:
> Hello,
>
<snipped for brevity>
> 2. We have 2 Single Number DDI numbers configured over the 4 channels.
> I want to set up outgoing calls so that they can appear to come from
> either of the two numbers, but at present the outgoing CLI appears to be
> overridden by the telco (BT) to only use one of the numbers.  Has anyone
> got this working in UK, and what is the format for the
> outbound_caller_id_number: last 6 digits or full 11 digits?  I note that
> the inbound called number is only the last 6 digits.
>
<snipped for brevity>
>
> Thanks for all the great support I have already got just from editing my
> Wiki User page; this is a friendly group!
>
> John
>

I can only comment on the BT part.

If they are two individual lines, you're out of luck with BT, as they
do not allow CLI spoofing, which this would be.

I would suggest sending BT the same number of digits that they send
you, if that doesn't work try 10 digits. If  you are not sending the
"correct" number of digits for a valid DDI on the trunk, then they
will send the lead number for the trunk.

Cheers


From anthony.minessale at gmail.com  Thu Jun  9 19:53:17 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 9 Jun 2011 10:53:17 -0500
Subject: [Freeswitch-users] Logical OR in conditions
In-Reply-To: <19EE2626-06B6-4971-BFCA-0370EF77A8AC@mralston.com>
References: <19EE2626-06B6-4971-BFCA-0370EF77A8AC@mralston.com>
Message-ID: <BANLkTi=FHz-3RYxQDp7W+LwhV7DYOGeQCg@mail.gmail.com>

the easy way would be to put the OR in the REGEX sep with |
^1234$|^5678$




On Thu, Jun 9, 2011 at 10:20 AM, Matthew Ralston
<freeswitch at mralston.com> wrote:
> Hi,
>
> What is the correct condition->break value to use in order to create a logical OR with conditions?
>
> I have two extensions which I'd like to merge into one. I'd like the actions to be run if either of the conditions matches:
>
> ? ? ? ? ? ? ? ?<extension name="call_forward_enable">
> ? ? ? ? ? ? ? ? ? ? ? ?<condition field="destination_number" expression="^\*21\*([0-9]+)#$">
> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?<action application="javascript" data="callForwardEnable.js"/>
> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?<action application="hangup"/>
> ? ? ? ? ? ? ? ? ? ? ? ?</condition>
> ? ? ? ? ? ? ? ?</extension>
>
> ? ? ? ? ? ? ? ?<!-- For phones that won't dial *# numbers, *=7, #=9 -->
> ? ? ? ? ? ? ? ?<extension name="call_forward_enable">
> ? ? ? ? ? ? ? ? ? ? ? ?<condition field="destination_number" expression="^7217([0-9]+)9$">
> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?<action application="javascript" data="callForwardEnable.js"/>
> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?<action application="hangup"/>
> ? ? ? ? ? ? ? ? ? ? ? ?</condition>
> ? ? ? ? ? ? ? ?</extension>
>
> So the result would be something like:
>
> ? ? ? ? ? ? ? ?<extension name="call_forward_enable">
> ? ? ? ? ? ? ? ? ? ? ? ?<condition field="destination_number" expression="^\*21\*([0-9]+)#$" break="on-true" />
> ? ? ? ? ? ? ? ? ? ? ? ?<condition field="destination_number" expression="^7217([0-9]+)9$">
> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?<action application="javascript" data="callForwardEnable.js"/>
> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?<action application="hangup"/>
> ? ? ? ? ? ? ? ? ? ? ? ?</condition>
> ? ? ? ? ? ? ? ?</extension>
>
> I can't seem to get it to work and my brain is melting trying to wrap my head around the different break values.
>
> Cheers,
>
> Matt
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From freeswitch at mralston.com  Thu Jun  9 20:10:08 2011
From: freeswitch at mralston.com (Matthew Ralston)
Date: Thu, 9 Jun 2011 17:10:08 +0100
Subject: [Freeswitch-users] Logical OR in conditions
In-Reply-To: <BANLkTi=FHz-3RYxQDp7W+LwhV7DYOGeQCg@mail.gmail.com>
References: <19EE2626-06B6-4971-BFCA-0370EF77A8AC@mralston.com>
	<BANLkTi=FHz-3RYxQDp7W+LwhV7DYOGeQCg@mail.gmail.com>
Message-ID: <31096DDC-7826-4038-8A27-D3B465AA4486@mralston.com>

Hi,

Ah ok, that'll do me. Thank you.

Kind regards,

Matthew Ralston
Web Developer & IT Consultant

matt at mralston.co.uk
www.mralston.com

On 9 Jun 2011, at 16:53, Anthony Minessale wrote:

> the easy way would be to put the OR in the REGEX sep with |
> ^1234$|^5678$
> 
> 
> 
> 
> On Thu, Jun 9, 2011 at 10:20 AM, Matthew Ralston
> <freeswitch at mralston.com> wrote:
>> Hi,
>> 
>> What is the correct condition->break value to use in order to create a logical OR with conditions?
>> 
>> I have two extensions which I'd like to merge into one. I'd like the actions to be run if either of the conditions matches:
>> 
>>               <extension name="call_forward_enable">
>>                       <condition field="destination_number" expression="^\*21\*([0-9]+)#$">
>>                               <action application="javascript" data="callForwardEnable.js"/>
>>                               <action application="hangup"/>
>>                       </condition>
>>               </extension>
>> 
>>               <!-- For phones that won't dial *# numbers, *=7, #=9 -->
>>               <extension name="call_forward_enable">
>>                       <condition field="destination_number" expression="^7217([0-9]+)9$">
>>                               <action application="javascript" data="callForwardEnable.js"/>
>>                               <action application="hangup"/>
>>                       </condition>
>>               </extension>
>> 
>> So the result would be something like:
>> 
>>               <extension name="call_forward_enable">
>>                       <condition field="destination_number" expression="^\*21\*([0-9]+)#$" break="on-true" />
>>                       <condition field="destination_number" expression="^7217([0-9]+)9$">
>>                               <action application="javascript" data="callForwardEnable.js"/>
>>                               <action application="hangup"/>
>>                       </condition>
>>               </extension>
>> 
>> I can't seem to get it to work and my brain is melting trying to wrap my head around the different break values.
>> 
>> Cheers,
>> 
>> Matt
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From msc at freeswitch.org  Thu Jun  9 20:37:41 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 9 Jun 2011 09:37:41 -0700
Subject: [Freeswitch-users] Enumerate variables in a session
In-Reply-To: <BANLkTikMT8GROmAdOnv8vBhuEEBYmiAeYw@mail.gmail.com>
References: <33095823FD21DF429B481B5163264B7950AC586589@VMBX102.ihostexchange.net>
	<BANLkTikmrXscsfdPUHhsgewYeF3RYNcyRQ@mail.gmail.com>
	<BANLkTikMT8GROmAdOnv8vBhuEEBYmiAeYw@mail.gmail.com>
Message-ID: <BANLkTimRZU7zdwCShvk83bW84=MtJppN7g@mail.gmail.com>

You can also create an API object and issue a uuid_dump <uuid> command and
parse the results.
-MC

On Thu, Jun 9, 2011 at 7:20 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Actually Christian, it is since you have a session object in javascript and
> have to access the variables via the Session api.
>
> The simple answer is not directly AFAIK, although I think there is an
> undocumented Session.getXMLCDR() function. The XML CDR contains all
> variables on the session, so if you parse that XML I think you'll get the
> full list.
>
> -Steve
>
>
>
> On 9 June 2011 15:03, Christian Benke <benkokakao at gmail.com> wrote:
>
>> On 8 June 2011 20:18, Yungwei Chen <yungwei at resolvity.com> wrote:
>> > Hi,
>> >
>> > I'm wondering if there's way to enumerate the list of variable names in
>> a session in javascript. Thanks.
>>
>> This is not a FreeSWITCH specific question - a javascript-focused
>> community or a more general programming website like stackoverflow.com
>> would be suited better.
>>
>> Best regards,
>> Christian
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From joland at gmail.com  Thu Jun  9 02:27:46 2011
From: joland at gmail.com (Jeppe Oland)
Date: Wed, 8 Jun 2011 22:27:46 +0000 (UTC)
Subject: [Freeswitch-users] Problem receiving calls from E1 to TE122p
References: <20110527195856.GA3362@jgalaz-desktop>
Message-ID: <loom.20110609T002558-842@post.gmane.org>

Javier Galaz Jeria <jgalaz at ...> writes:

> I'm using FS git, on Ubuntu 11.04, with Digium TE122p card (from now on
> srv1), and FS git, on Ubuntu Server 11.04 with Sangoma A101 card (from now
> on srv2), both currently configured by vendor's scripts.

I was trying to get a Sangoma A200 working on Ubuntu 11.04 last week, but as far
as I can tell, WANPIPE won't compile with the new kernels.

What did you do to get it working in the first place?

Regards,
-Jeppe



From kahn at vestec.com  Thu Jun  9 20:15:01 2011
From: kahn at vestec.com (Kashif Kahn)
Date: Thu, 09 Jun 2011 12:15:01 -0400
Subject: [Freeswitch-users] Speech Recognition: Chinese Mandarin
Message-ID: <4DF0F185.8010500@vestec.com>

Hello Everyone,

Vestec ASR engine now supports speech recognition in Chinese Mandarin. 
(We also support American English, Australian English, and Indian 
English). A starter kit is available for $25: http://www.vestec.com/products

Please note that Vestec offers the best deal around for enabling 
sophisticated speech recognition with command-and-control type IVR 
applications by offering a high accuracy, standards based speech engine 
at a fraction of the cost of conventional ASR vendors. Feel free to 
contact me with any questions or concerns.

Regards,
-Kashif



From msc at freeswitch.org  Thu Jun  9 20:47:28 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 9 Jun 2011 09:47:28 -0700
Subject: [Freeswitch-users] originating a call from an outbound socket
	app
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD021F45EAA0@NY1-EXMB-01.ip-soft.net>
References: <6A6B4C284AD15042B429EB9D904544AD021F45EAA0@NY1-EXMB-01.ip-soft.net>
Message-ID: <BANLkTi=LX=cGqSVrT6o-0OOPXPOA2U=MnA@mail.gmail.com>

I would say that you want to bridge the inbound leg to the outbound leg and
use the answer confirmation variables. I recommend you pastebin a simple
java app that answers the call and attempts to bridge it and also pastebin
the console log. The community will take a look and see if we can help you
figure out what is happening.

-MC

On Wed, Jun 8, 2011 at 3:14 PM, Hector Geraldino <
Hector.Geraldino at ip-soft.net> wrote:

> Hi all,
>
>
>
> I?ve been playing with FS around weeks, with some level of success. Right
> now I?m stuck trying to dial an extension, or to bridge a call to an
> external extension. Briefly, the scenario is the following:
>
>
>
> -          Cisco CCM forwarding SIP calls to FS
>
> -          FS in outbound mode,  forwarding calls to a Java application
> using freeswitch-esl jar
>
>
>
> The java application answers the call, says a dialogue using TTS and
> receives input using DTFM or speech commands; all this is working pretty
> well. Now I?m trying to put this call in hold, originate a call to another
> extension and, depending on the result (call answered, not answered,
> destination user presses an specific number generating an expected DTMF), I
> need to bridge both calls and finish the session.
>
>
>
> I try several ways to get this done with no luck yet: using api originate
> to make the call with different arguments ({ignore_early_media,  etc). I
> even tried to skip this step and just do the bridge between the inbound call
> and the external extension, but this  didn?t work either. I can?t see any
> errors thrown by FS, and the result of the command is +OK, but nothing
> happens, the external phone never rings. The FS log only says:
>
>
>
> [DEBUG] switch_core_session.c:954 Send signal sofia/external/
> 5655 at 192.168.8.1 [BREAK] <= this ext/IP address is from the inbound
> caller, not the one I?m trying to connect to.
>
>
>
> My first impression was that I might have something wrong with the
> originate/bridge parameters. I changed the dialplan XML file to just do the
> bridge, and it worked like a charm.
>
>
>
> <include>
>
> <extension name="test">
>
>         <condition field="destination_number" expression="6114|5121">
>
> <!--    <action application="socket" data="localhost:8044 async full"/>
> -->
>
>           <action application="bridge" data="sofia/external/
> 5562 at 192.168.2.14" />
>
>         </condition>
>
> </extension>
>
> </include>
>
>
>
> I don?t know what path to follow here. I was unable to find relevant
> examples on google that guides me to solve this little issue. Any advices?
>
>
>
> Thanks all,
>
> Hector
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From robert.hadley at teotech.com  Thu Jun  9 21:43:01 2011
From: robert.hadley at teotech.com (Robert Hadley)
Date: Thu, 9 Jun 2011 10:43:01 -0700
Subject: [Freeswitch-users] Voicemail Message - How t know how many
	are	read & how many are unread in Inbox
In-Reply-To: <BANLkTikshbtzM67daHBePp7TB5pH5FjEyA@mail.gmail.com>
References: <mailman.10243.1307524956.24555.freeswitch-users@lists.freeswitch.org>
	<AA1BE0FA88F3421784DBD1D91A84E515@sharad>
	<BANLkTikshbtzM67daHBePp7TB5pH5FjEyA@mail.gmail.com>
Message-ID: <E826C94A3C6BE44198FAF4D66B5AB4D81BDA4D5BF1@VA3DIAXVS351.RED001.local>


vm_boxcount user at domain|[new|saved|new-urgent|saved-urgent|all]]

freeswitch at internal> vm_boxcount 2002 at 192.168.xx.xx|all
5:0:0:0

-Robert

-----Original Message-----
From: Christian Benke [mailto:benkokakao at gmail.com] 
Sent: Thursday, June 09, 2011 7:33 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Voicemail Message - How t know how many are read & how many are unread in Inbox

> Just wondering whether there is any way to know how many voice 
> messages are read or unread in New & SAVED box.

Not sure in "which" way you want to know this. From fs_cli you can try:

> vm_list user at domain xml

This gives you an xml-output of all messages of this user - saved/read mails have an read_epoch-value, unread messages have a read_epoch-value of 0, e.g.:

freeswitch at internal> vm_list 200 at 192.168.0.4 xml <voicemail>  <message>
  <created_epoch>1307630009</created_epoch>
  <read_epoch>0</read_epoch>
  <username>200</username>
  <domain>192.168.0.4</domain>
  <folder>inbox</folder>
  <path>/usr/local/freeswitch/storage/voicemail/default/192.168.0.4/200/msg_60c27a70-92a5-11e0-8b2f-fd7505ade349.wav</path>
  <uuid>60c27a70-92a5-11e0-8b2f-fd7505ade349</uuid>
  <cid-name>Jim Doe</cid-name>
  <cid-number>300</cid-number>
 </message>
 <message>
  <created_epoch>1307348316</created_epoch>
  <read_epoch>1307629941</read_epoch>
  <username>200</username>
  <domain>192.168.0.4</domain>
  <folder>inbox</folder>
  <path>/usr/local/freeswitch/storage/voicemail/default/192.168.0.4/200/msg_775fa510-9015-11e0-88e7-fd7505ade349.wav</path>
  <uuid>775fa510-9015-11e0-88e7-fd7505ade349</uuid>
  <cid-name>John Doe</cid-name>
  <cid-number>100</cid-number>
 </message>
</voicemail>

freeswitch at internal>

hth
Christian



From freeswitch at earthspike.net  Thu Jun  9 22:10:11 2011
From: freeswitch at earthspike.net (John)
Date: Thu, 09 Jun 2011 19:10:11 +0100
Subject: [Freeswitch-users] UK ISDN outgoing CLI (was Re: FreeTDM - Sangoma
 B700 - ISDN connection questions - UK)
In-Reply-To: <BANLkTimP23FVs8Vcg+pbxnfBucrQoJur2w@mail.gmail.com>
References: <4DED4854.6010203@earthspike.net>
	<BANLkTimP23FVs8Vcg+pbxnfBucrQoJur2w@mail.gmail.com>
Message-ID: <4DF10C83.6020803@earthspike.net>

[I've forked this thread as it really should have been 3 separate 
threads to start with.]

Thanks,

I have tried every combination I can think of, except 10 digits which I 
will give a go.  That makes sense as in the general case it would permit 
non-geographical numbers to be presented.

The lines are provided as 2 x ISDN2e with a lead number covering all 4 
channels.  The NTE8s are labelled 1-2 and 3-4, so not separate 
individual lines, I thnk.

We currently have a Single Number DDI in addition, and hope to increase 
that to 2 very soon; the 3 numbers between them will serve 3 separate 
business units, so we want to be able to present the correct outbound 
CLI from each business unit.  We also have a remote call forward from 
another number to the lead number, but presenting that really would be 
CLI spoofing, so I am not expecting that to be permissible. But I would 
expect those that are directly allocated to the 2 lines/4 channels to be 
allowed to be presented.

John

On 09/06/11 08:27, shouldbe q931 wrote:
> On Mon, Jun 6, 2011 at 10:36 PM, John<freeswitch at earthspike.net>  wrote:
>> Hello,
>>
> <snipped for brevity>
>> 2. We have 2 Single Number DDI numbers configured over the 4 channels.
>> I want to set up outgoing calls so that they can appear to come from
>> either of the two numbers, but at present the outgoing CLI appears to be
>> overridden by the telco (BT) to only use one of the numbers.  Has anyone
>> got this working in UK, and what is the format for the
>> outbound_caller_id_number: last 6 digits or full 11 digits?  I note that
>> the inbound called number is only the last 6 digits.
>>
> <snipped for brevity>
>> Thanks for all the great support I have already got just from editing my
>> Wiki User page; this is a friendly group!
>>
>> John
>>
> I can only comment on the BT part.
>
> If they are two individual lines, you're out of luck with BT, as they
> do not allow CLI spoofing, which this would be.
>
> I would suggest sending BT the same number of digits that they send
> you, if that doesn't work try 10 digits. If  you are not sending the
> "correct" number of digits for a valid DDI on the trunk, then they
> will send the lead number for the trunk.
>
> Cheers
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From Hector.Geraldino at ip-soft.net  Fri Jun 10 00:22:25 2011
From: Hector.Geraldino at ip-soft.net (Hector Geraldino)
Date: Thu, 9 Jun 2011 16:22:25 -0400
Subject: [Freeswitch-users] originating a call from an outbound socket
 app
In-Reply-To: <BANLkTi=LX=cGqSVrT6o-0OOPXPOA2U=MnA@mail.gmail.com>
References: <6A6B4C284AD15042B429EB9D904544AD021F45EAA0@NY1-EXMB-01.ip-soft.net>
	<BANLkTi=LX=cGqSVrT6o-0OOPXPOA2U=MnA@mail.gmail.com>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD021F45EB71@NY1-EXMB-01.ip-soft.net>

Hi Michael,

Thanks for replying back. I've pasted a sample application on pastebin, the URL is:

http://pastebin.com/MYusWe3V

What I'm trying to do (and have partially accomplished) is:

-Answer the call
- Play a message
- wait for a command (dtmf or voice, this is out of the sample app for the sake of keeping it short & simple)
- originate a new call, wait for it to be answered and depending on an event (dtmf)...
- ...do the bridge or dial another extension.

The extension number is external (managed by a Cisco CCM). If I try to do the bridge from the XML dialplan it works perfect. When I try to do it from an outbound socket java application, anything happens, and almost no information is logged on FS log that I can use to decide what might be happening.

Thanks again for your help. I truly appreciate it.

Hector

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Thursday, June 09, 2011 12:47 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] originating a call from an outbound socket app

I would say that you want to bridge the inbound leg to the outbound leg and use the answer confirmation variables. I recommend you pastebin a simple java app that answers the call and attempts to bridge it and also pastebin the console log. The community will take a look and see if we can help you figure out what is happening.

-MC
On Wed, Jun 8, 2011 at 3:14 PM, Hector Geraldino <Hector.Geraldino at ip-soft.net<mailto:Hector.Geraldino at ip-soft.net>> wrote:
Hi all,

I've been playing with FS around weeks, with some level of success. Right now I'm stuck trying to dial an extension, or to bridge a call to an external extension. Briefly, the scenario is the following:


-          Cisco CCM forwarding SIP calls to FS

-          FS in outbound mode,  forwarding calls to a Java application using freeswitch-esl jar

The java application answers the call, says a dialogue using TTS and receives input using DTFM or speech commands; all this is working pretty well. Now I'm trying to put this call in hold, originate a call to another extension and, depending on the result (call answered, not answered, destination user presses an specific number generating an expected DTMF), I need to bridge both calls and finish the session.

I try several ways to get this done with no luck yet: using api originate to make the call with different arguments ({ignore_early_media,  etc). I even tried to skip this step and just do the bridge between the inbound call and the external extension, but this  didn't work either. I can't see any errors thrown by FS, and the result of the command is +OK, but nothing happens, the external phone never rings. The FS log only says:

[DEBUG] switch_core_session.c:954 Send signal sofia/external/5655 at 192.168.8.1<mailto:5655 at 192.168.8.1> [BREAK] <= this ext/IP address is from the inbound caller, not the one I'm trying to connect to.

My first impression was that I might have something wrong with the originate/bridge parameters. I changed the dialplan XML file to just do the bridge, and it worked like a charm.

<include>
<extension name="test">
        <condition field="destination_number" expression="6114|5121">
<!--    <action application="socket" data="localhost:8044 async full"/>  -->
          <action application="bridge" data="sofia/external/5562 at 192.168.2.14<mailto:5562 at 192.168.2.14>" />
        </condition>
</extension>
</include>

I don't know what path to follow here. I was unable to find relevant examples on google that guides me to solve this little issue. Any advices?

Thanks all,
Hector

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FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
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From kheimerl at cs.berkeley.edu  Fri Jun 10 00:21:59 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Thu, 9 Jun 2011 13:21:59 -0700
Subject: [Freeswitch-users] Sofia invite issues
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59DEE58301@cooper>
References: <BANLkTikWTD7bT+kiMdDk1VTOU2XsPU0rcQ@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DEE58301@cooper>
Message-ID: <BANLkTinvK0rhcqov3SL8aCqp8UL4sJ9Y_Q@mail.gmail.com>

Here you go: http://pastebin.freeswitch.org/16468

For this packet trace I decided to have profile "C" in the UA
directory. This causes FS to send the second invite (which should be
A->B) as C->B, rather than B->B, as it does if that UA is not present.
 Neither is correct, this just makes it a little easier to grep.

User 1302 first appears in that invite, I don't see any other log
items talking about 1302 at all. It's worth noting that 1302's user
name (IMSI641104878332498) DOES appear in the SDP packet loaded in the
first (A->B) invite, but I'm pretty certain FS should not be using
that information for SIP addressing. Again, if I remove user
IMSI641104878332498 from the directory, FS just switched to returning
with user 1301, it doesn't fix the issue.

It's very likely I'm massaging some code that's not commonly used, as
I may be making a small mistake here or there in the original SIP
invite. These two lines:
2011-06-09 13:11:32.947157 [DEBUG] sofia.c:6674 IP 192.168.1.144
Approved by acl "domains[IMSI641104878332498 at 192.168.1.144]". Access
Granted.
2011-06-09 13:11:32.947157 [DEBUG] sofia.c:6803 Authenticating user
IMSI641104878332498 at 192.168.1.144
Are particularly curious. I'm not sure if those are a response to the
incoming invite (which would be incorrect) or not. It should be
approving user 1300 at 192.168.1.144, if anyone.

Thanks!

On Thu, Jun 9, 2011 at 12:18 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> Please pastebin a complete debug log, with "sofia global siptrace on", to http://pastebin.freeswitch.org.
>
> I'm pretty sure this is not a bug in FS, if it was, we would have lots of people complaining...
>
> /Peter
>
>
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Kurtis Heimerl
> Skickat: den 8 juni 2011 23:32
> Till: freeswitch-users at lists.freeswitch.org
> ?mne: [Freeswitch-users] Sofia invite issues
>
> Hello Freeswitch-users!
>
> I'm currently writing a SIP middlebox that intercepts sip messages and
> changes their username. The reasons for this are complicated and
> somewhat outside of the scope of this discussion. However, I've come
> upon a very strange issue: when making a phone-to-phone call across
> freeswitch; it is connecting the wrong user. I'm fairly well convinced
> this is a bug, but I thought I'd send the issue here and see if it's
> anything I'm obviously doing wrong.
>
> Basically, FS sees a SIP message from a registered UA (call it A)
> inviting another registered user to a call (B). This is acked
> correctly (a TRYING message). FS then responds by inviting B into a
> call with some OTHER user C. C is not mentioned at any point in the
> initial SIP messages (as verified by wireshark). Even more strangely,
> if I remove user C from FS (by removing their config file) FS responds
> to the invite by inviting B into a call with itself. In each case, the
> appropriate from header in the invite should be the original caller A.
> ?It's worth noting that I am not monkeying with the RTP packets at
> all, but my understanding is that SIP signalling shouldn't be affected
> by that.
>
> My general guess is that I'm messing up the naming somehow, and FS is
> running an algorithm to guess at who the call originator is. However,
> the naming must be roughly correct; Asterisk is able to handle this
> call just fine. I've included a sip trace of the second situation (A
> calls B, FS invites B to a call with itself)
>
> For the record A:1300
> B:1301
> C:IMSI641104878332498
>
> REGISTER MESSAGE (Which works fine)
>
> REGISTER sip:192.168.1.144 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
> From: 1300 <sip:1300 at 192.168.1.144>;tag=ftoui
> To: 1300 <sip:1300 at 192.168.1.144>
> Call-ID: 1032827938 at 192.168.1.144
> CSeq: 91 REGISTER
> Contact: <sip:1300 at 192.168.1.144:5063>;expires=7200
> Max-Forwards: 70
> Content-Length: 0
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
> From: 1300 <sip:1300 at 192.168.1.144>;tag=ftoui
> To: 1300 <sip:1300 at 192.168.1.144>;tag=jU64NXypQc57F
> Call-ID: 1032827938 at 192.168.1.144
> CSeq: 91 REGISTER
> Contact: <sip:1300 at 192.168.1.144:5063>;expires=7200
> Date: Wed, 08 Jun 2011 21:02:04 GMT
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-71d64e6 2011-06-07 22-30-16 -0700
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Content-Length: 0
>
> SIP TRACE (A -> B, B -> B response)
>
> INITIAL INVITE
> INVITE sip:1301 at 192.168.1.144 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK77974
> From: 1300 <sip:1300 at 192.168.1.144>;tag=bgdqx
> To: <sip:1301 at 192.168.1.144>
> Call-ID: 1817795092 at 192.168.1.144
> CSeq: 485 INVITE
> Contact: <sip:1300 at 192.168.1.144:5063>;expires=3600
> Content-Type: application/sdp
> Max-Forwards: 70
> Content-Length: 143
>
> INVITE ACK
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK77974
> From: 1300 <sip:1300 at 192.168.1.144>;tag=bgdqx
> To: <sip:1301 at 192.168.1.144>
> Call-ID: 1817795092 at 192.168.1.144
> CSeq: 485 INVITE
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-71d64e6 2011-06-07 22-30-16 -0700
> Content-Length: 0
>
> FS INVITE (Note the from address being wrong)
> INVITE sip:1301 at 192.168.1.144:5063 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.144;rport;branch=z9hG4bKeeQvcyZ70SDUg
> Max-Forwards: 69
> From: "1301" <sip:1301 at 192.168.1.144>;tag=rHyS0Z3B61arN
> To: <sip:1301 at 192.168.1.144:5063>
> Call-ID: 84df47f9-0cb7-122f-13b5-5cff350d9de5
> CSeq: 13447852 INVITE
> Contact: <sip:mod_sofia at 192.168.1.144:5060>
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-71d64e6 2011-06-07 22-30-16 -0700
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
> sla, include-session-description, presence.winfo, message-summary,
> refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 315
> X-FS-Support: update_display
> Remote-Party-ID: "1301"
> <sip:1301 at 192.168.1.144>;party=calling;screen=yes;privacy=off
>
> USER CONFIGURATIONS
>
> 1300.xml
> <include>
> ?<user id="1300" cidr="192.168.1.0/24,127.0.0.1/32">
> ? ?<params>
> ? ? ?<param name="accept-blind-reg" value="true"/>
> ? ? ?<param name="accept-blind-auth" value="true"/>
> ? ? ?<param name="vm-password" value="1000"/>
> ? ?</params>
> ? ?<variables>
> ? ? ?<variable name="toll_allow" value="domestic,international,local"/>
> ? ? ?<variable name="accountcode" value="1300"/>
> ? ? ?<variable name="user_context" value="default"/>
> ? ? ?<variable name="effective_caller_id_name" value="1300"/>
> ? ? ?<variable name="effective_caller_id_number" value="1300"/>
> ? ? ?<variable name="outbound_caller_id_name" value="1300"/>
> ? ? ?<variable name="outbound_caller_id_number" value="1300"/>
> ? ? ?<variable name="callgroup" value="techsupport"/>
> ? ?</variables>
> ?</user>
> </include>
>
> 1301.xml
> <include>
> ?<user id="1301" cidr="192.168.1.0/24,127.0.0.1/32">
> ? ?<params>
> ? ? ?<param name="accept-blind-reg" value="true"/>
> ? ? ?<param name="accept-blind-auth" value="true"/>
> ? ? ?<param name="vm-password" value="1000"/>
> ? ?</params>
> ? ?<variables>
> ? ? ?<variable name="toll_allow" value="domestic,international,local"/>
> ? ? ?<variable name="accountcode" value="1301"/>
> ? ? ?<variable name="user_context" value="default"/>
> ? ? ?<variable name="effective_caller_id_name" value="1301"/>
> ? ? ?<variable name="effective_caller_id_number" value="1301"/>
> ? ? ?<variable name="outbound_caller_id_name" value="1301"/>
> ? ? ?<variable name="outbound_caller_id_number" value="1301"/>
> ? ? ?<variable name="callgroup" value="techsupport"/>
> ? ?</variables>
> ?</user>
> </include>
>
> IMSI641104878332498.xml
> <include>
> ?<user id="IMSI641104878332498" cidr="192.168.1.0/24,127.0.0.1/32"
> number-alias="1302">
> ? ?<params>
> ? ? ?<param name="accept-blind-reg" value="true"/>
> ? ? ?<param name="accept-blind-auth" value="true"/>
> ? ? ?<param name="vm-password" value="1000"/>
> ? ?</params>
> ? ?<variables>
> ? ? ?<variable name="toll_allow" value="domestic,international,local"/>
> ? ? ?<variable name="accountcode" value="1302"/>
> ? ? ?<variable name="user_context" value="default"/>
> ? ? ?<variable name="effective_caller_id_name" value="IMSI641104878332498"/>
> ? ? ?<variable name="effective_caller_id_number" value="1302"/>
> ? ? ?<variable name="outbound_caller_id_name" value="IMSI641104878332498"/>
> ? ? ?<variable name="outbound_caller_id_number" value="1302"/>
> ? ? ?<variable name="callgroup" value="techsupport"/>
> ? ?</variables>
> ?</user>
> </include>
>
> DIALPLAN
>
> ? ?<extension name="Local Numbers">
> ? ? ?<condition field="destination_number" expression="^130[0-9]$">
> ? ? ? ?<!--<action application="bridge"
> data="user/${user_data(${destination_number}@${domain_name} attr
> id)}"/> -->
> ? ? ? ?<action application="bridge" data="user/${destination_number}"/>
> ? ? ?</condition>
> ? ?</extension>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> !DSPAM:4defeb7432764293448790!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From peter.olsson at visionutveckling.se  Fri Jun 10 00:55:22 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 9 Jun 2011 22:55:22 +0200
Subject: [Freeswitch-users] Sofia invite issues
In-Reply-To: <BANLkTinvK0rhcqov3SL8aCqp8UL4sJ9Y_Q@mail.gmail.com>
References: <BANLkTikWTD7bT+kiMdDk1VTOU2XsPU0rcQ@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DEE58301@cooper>,
	<BANLkTinvK0rhcqov3SL8aCqp8UL4sJ9Y_Q@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59DEF56421@cooper>

I'm guessing your problem might be related to the fact that you use the cidr attribute on the user's - with the same IP's for all of them. If I understand these correctly (never used it this way myself) they're supposed to be used to authenticate a user for a specific IP, instead of challenge auth, and since the same IP is provided for all of them they might overwrite eachother, and the last one will be the one it matches.

Also, read more on http://wiki.freeswitch.org/wiki/Acl#Users.

Try removing the cidr attribute, and let the agents auth instead, or use different IP's for the users.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Kurtis Heimerl [kheimerl at cs.berkeley.edu]
Skickat: den 9 juni 2011 22:21
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Sofia invite issues

Here you go: http://pastebin.freeswitch.org/16468

For this packet trace I decided to have profile "C" in the UA
directory. This causes FS to send the second invite (which should be
A->B) as C->B, rather than B->B, as it does if that UA is not present.
 Neither is correct, this just makes it a little easier to grep.

User 1302 first appears in that invite, I don't see any other log
items talking about 1302 at all. It's worth noting that 1302's user
name (IMSI641104878332498) DOES appear in the SDP packet loaded in the
first (A->B) invite, but I'm pretty certain FS should not be using
that information for SIP addressing. Again, if I remove user
IMSI641104878332498 from the directory, FS just switched to returning
with user 1301, it doesn't fix the issue.

It's very likely I'm massaging some code that's not commonly used, as
I may be making a small mistake here or there in the original SIP
invite. These two lines:
2011-06-09 13:11:32.947157 [DEBUG] sofia.c:6674 IP 192.168.1.144
Approved by acl "domains[IMSI641104878332498 at 192.168.1.144]". Access
Granted.
2011-06-09 13:11:32.947157 [DEBUG] sofia.c:6803 Authenticating user
IMSI641104878332498 at 192.168.1.144
Are particularly curious. I'm not sure if those are a response to the
incoming invite (which would be incorrect) or not. It should be
approving user 1300 at 192.168.1.144, if anyone.

Thanks!

On Thu, Jun 9, 2011 at 12:18 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> Please pastebin a complete debug log, with "sofia global siptrace on", to http://pastebin.freeswitch.org.
>
> I'm pretty sure this is not a bug in FS, if it was, we would have lots of people complaining...
>
> /Peter
>
>
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Kurtis Heimerl
> Skickat: den 8 juni 2011 23:32
> Till: freeswitch-users at lists.freeswitch.org
> ?mne: [Freeswitch-users] Sofia invite issues
>
> Hello Freeswitch-users!
>
> I'm currently writing a SIP middlebox that intercepts sip messages and
> changes their username. The reasons for this are complicated and
> somewhat outside of the scope of this discussion. However, I've come
> upon a very strange issue: when making a phone-to-phone call across
> freeswitch; it is connecting the wrong user. I'm fairly well convinced
> this is a bug, but I thought I'd send the issue here and see if it's
> anything I'm obviously doing wrong.
>
> Basically, FS sees a SIP message from a registered UA (call it A)
> inviting another registered user to a call (B). This is acked
> correctly (a TRYING message). FS then responds by inviting B into a
> call with some OTHER user C. C is not mentioned at any point in the
> initial SIP messages (as verified by wireshark). Even more strangely,
> if I remove user C from FS (by removing their config file) FS responds
> to the invite by inviting B into a call with itself. In each case, the
> appropriate from header in the invite should be the original caller A.
>  It's worth noting that I am not monkeying with the RTP packets at
> all, but my understanding is that SIP signalling shouldn't be affected
> by that.
>
> My general guess is that I'm messing up the naming somehow, and FS is
> running an algorithm to guess at who the call originator is. However,
> the naming must be roughly correct; Asterisk is able to handle this
> call just fine. I've included a sip trace of the second situation (A
> calls B, FS invites B to a call with itself)
>
> For the record A:1300
> B:1301
> C:IMSI641104878332498
>
> REGISTER MESSAGE (Which works fine)
>
> REGISTER sip:192.168.1.144 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
> From: 1300 <sip:1300 at 192.168.1.144>;tag=ftoui
> To: 1300 <sip:1300 at 192.168.1.144>
> Call-ID: 1032827938 at 192.168.1.144
> CSeq: 91 REGISTER
> Contact: <sip:1300 at 192.168.1.144:5063>;expires=7200
> Max-Forwards: 70
> Content-Length: 0
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
> From: 1300 <sip:1300 at 192.168.1.144>;tag=ftoui
> To: 1300 <sip:1300 at 192.168.1.144>;tag=jU64NXypQc57F
> Call-ID: 1032827938 at 192.168.1.144
> CSeq: 91 REGISTER
> Contact: <sip:1300 at 192.168.1.144:5063>;expires=7200
> Date: Wed, 08 Jun 2011 21:02:04 GMT
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-71d64e6 2011-06-07 22-30-16 -0700
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Content-Length: 0
>
> SIP TRACE (A -> B, B -> B response)
>
> INITIAL INVITE
> INVITE sip:1301 at 192.168.1.144 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK77974
> From: 1300 <sip:1300 at 192.168.1.144>;tag=bgdqx
> To: <sip:1301 at 192.168.1.144>
> Call-ID: 1817795092 at 192.168.1.144
> CSeq: 485 INVITE
> Contact: <sip:1300 at 192.168.1.144:5063>;expires=3600
> Content-Type: application/sdp
> Max-Forwards: 70
> Content-Length: 143
>
> INVITE ACK
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK77974
> From: 1300 <sip:1300 at 192.168.1.144>;tag=bgdqx
> To: <sip:1301 at 192.168.1.144>
> Call-ID: 1817795092 at 192.168.1.144
> CSeq: 485 INVITE
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-71d64e6 2011-06-07 22-30-16 -0700
> Content-Length: 0
>
> FS INVITE (Note the from address being wrong)
> INVITE sip:1301 at 192.168.1.144:5063 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.144;rport;branch=z9hG4bKeeQvcyZ70SDUg
> Max-Forwards: 69
> From: "1301" <sip:1301 at 192.168.1.144>;tag=rHyS0Z3B61arN
> To: <sip:1301 at 192.168.1.144:5063>
> Call-ID: 84df47f9-0cb7-122f-13b5-5cff350d9de5
> CSeq: 13447852 INVITE
> Contact: <sip:mod_sofia at 192.168.1.144:5060>
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-71d64e6 2011-06-07 22-30-16 -0700
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
> sla, include-session-description, presence.winfo, message-summary,
> refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 315
> X-FS-Support: update_display
> Remote-Party-ID: "1301"
> <sip:1301 at 192.168.1.144>;party=calling;screen=yes;privacy=off
>
> USER CONFIGURATIONS
>
> 1300.xml
> <include>
>  <user id="1300" cidr="192.168.1.0/24,127.0.0.1/32">
>    <params>
>      <param name="accept-blind-reg" value="true"/>
>      <param name="accept-blind-auth" value="true"/>
>      <param name="vm-password" value="1000"/>
>    </params>
>    <variables>
>      <variable name="toll_allow" value="domestic,international,local"/>
>      <variable name="accountcode" value="1300"/>
>      <variable name="user_context" value="default"/>
>      <variable name="effective_caller_id_name" value="1300"/>
>      <variable name="effective_caller_id_number" value="1300"/>
>      <variable name="outbound_caller_id_name" value="1300"/>
>      <variable name="outbound_caller_id_number" value="1300"/>
>      <variable name="callgroup" value="techsupport"/>
>    </variables>
>  </user>
> </include>
>
> 1301.xml
> <include>
>  <user id="1301" cidr="192.168.1.0/24,127.0.0.1/32">
>    <params>
>      <param name="accept-blind-reg" value="true"/>
>      <param name="accept-blind-auth" value="true"/>
>      <param name="vm-password" value="1000"/>
>    </params>
>    <variables>
>      <variable name="toll_allow" value="domestic,international,local"/>
>      <variable name="accountcode" value="1301"/>
>      <variable name="user_context" value="default"/>
>      <variable name="effective_caller_id_name" value="1301"/>
>      <variable name="effective_caller_id_number" value="1301"/>
>      <variable name="outbound_caller_id_name" value="1301"/>
>      <variable name="outbound_caller_id_number" value="1301"/>
>      <variable name="callgroup" value="techsupport"/>
>    </variables>
>  </user>
> </include>
>
> IMSI641104878332498.xml
> <include>
>  <user id="IMSI641104878332498" cidr="192.168.1.0/24,127.0.0.1/32"
> number-alias="1302">
>    <params>
>      <param name="accept-blind-reg" value="true"/>
>      <param name="accept-blind-auth" value="true"/>
>      <param name="vm-password" value="1000"/>
>    </params>
>    <variables>
>      <variable name="toll_allow" value="domestic,international,local"/>
>      <variable name="accountcode" value="1302"/>
>      <variable name="user_context" value="default"/>
>      <variable name="effective_caller_id_name" value="IMSI641104878332498"/>
>      <variable name="effective_caller_id_number" value="1302"/>
>      <variable name="outbound_caller_id_name" value="IMSI641104878332498"/>
>      <variable name="outbound_caller_id_number" value="1302"/>
>      <variable name="callgroup" value="techsupport"/>
>    </variables>
>  </user>
> </include>
>
> DIALPLAN
>
>    <extension name="Local Numbers">
>      <condition field="destination_number" expression="^130[0-9]$">
>        <!--<action application="bridge"
> data="user/${user_data(${destination_number}@${domain_name} attr
> id)}"/> -->
>        <action application="bridge" data="user/${destination_number}"/>
>      </condition>
>    </extension>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

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

!DSPAM:4df12cbe32761899399581!



From chat2jesse at gmail.com  Fri Jun 10 01:30:38 2011
From: chat2jesse at gmail.com (jesse)
Date: Thu, 9 Jun 2011 14:30:38 -0700
Subject: [Freeswitch-users] how to redirect $${sounds_dir} ?
Message-ID: <BANLkTinv+-0wP0NPyAVTRsG595HPCmUObtTNfy8m0o7f7_Gspw@mail.gmail.com>

I would like to pint sounds_dir to a different location, but it seems
freeswitch doesn't support --sounds_dir  as command line arguments.
please let me know how to re-define sounds_dir?

thanks!
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From mehmasarja at gmail.com  Thu Jun  9 21:10:07 2011
From: mehmasarja at gmail.com (Mehma Sarja)
Date: Thu, 09 Jun 2011 10:10:07 -0700
Subject: [Freeswitch-users] Default Install Extensions NEWBIE Question
Message-ID: <4DF0FE6F.2020806@gmail.com>

The default install on ubuntu renders a working system servicing 19 
extensions. Can I use this as-is in a production environment which only 
requires within-LAN, inter-departmental voice connectivity?

Mehma


From ivank at rogers.com  Fri Jun 10 00:38:22 2011
From: ivank at rogers.com (Ivan Kovacevic)
Date: Thu, 9 Jun 2011 16:38:22 -0400
Subject: [Freeswitch-users] continue_on_fail failover with mod_lcr
Message-ID: <027301cc26e5$2d224a50$8766def0$@rogers.com>

Hi Everyone,

 

I would like to implement fail-over for my outbound gateway, but I would like to be able to pick certain cause codes which qualify
to stop trying next gateway (specifically when I have bad number and I am getting sip:404 NO_ROUTE_DESTINATION, UNALLOCATED_NUMBER).
I was using 1.0.6, but I moved to the newest git about month ago.

 

After searching through lists and spending several hours playing with continue_to_fail, failure_causes and fail_on_single_reject I
was able to make it work by specifying  cause codes for which I want to fail-over and omitting ones that qualify to stop trying next
gateway in variable continue_to_fail:

 

So this setup is working for me and in the case I have bad number and x.x.x.x returns sip:404 (NO_ROUTE_DESTINATION or
UNALLOCATED_NUMBER) it is not trying y.y.y.y gateway.

 

 

<include>

  <extension name="myextension">

    <condition field="destination_number" expression="^(01144\d+)$">

      <action application="set"

data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,SERVICE_NOT_IMPLEMENTED,CALL_RE

JECTED,NO_USER_RESPONSE,EXCHANGE_ROUTING_ERROR,INCOMPATIBLE_DESTINATION,NETWORK_OUT_OF_ORDER,RECOVERY_ON_TIMER_EXPIRE"/>

       <action application="bridge" data="sofia/myprofile/$1 at x.x.x.x"/>

       <action application="bridge" data="sofia/myprofile/$1 at y.y.y.y"/>

       <action application="bridge" data="sofia/myprofile/$1 at z.z.z.z"/>

    </condition>

  </extension>

</include>

 

 

However if I want to use "|" between my gateways - and the example below is not working. And no matter what x.x.x.x returns - it
will try y.y.y.y  and eventually z.z.z.z.

 

<action application="set"

data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,SERVICE_NOT_IMPLEMENTED,CALL_RE

JECTED,NO_USER_RESPONSE,EXCHANGE_ROUTING_ERROR,INCOMPATIBLE_DESTINATION,NETWORK_OUT_OF_ORDER,RECOVERY_ON_TIMER_EXPIRE"/>

<action application="bridge"

data="sofia/myprofile/$1 at x.x.x.x|sofia/myprofile/$1 at y.y.y.y|sofia/myprofile/$1 at z.z.z.z"/>

 

Unfortunately, we have to use pipe for fail-over since we are using mod_lcr to choose between outbound gateways.

 

Any suggestions? 

 

Thanks,

 

Ivan 

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From msc at freeswitch.org  Fri Jun 10 02:05:21 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 9 Jun 2011 15:05:21 -0700
Subject: [Freeswitch-users] originating a call from an outbound socket
	app
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD021F45EB71@NY1-EXMB-01.ip-soft.net>
References: <6A6B4C284AD15042B429EB9D904544AD021F45EAA0@NY1-EXMB-01.ip-soft.net>
	<BANLkTi=LX=cGqSVrT6o-0OOPXPOA2U=MnA@mail.gmail.com>
	<6A6B4C284AD15042B429EB9D904544AD021F45EB71@NY1-EXMB-01.ip-soft.net>
Message-ID: <BANLkTimszWvEvXbMagKnciMQ_ZzY=mfv3A@mail.gmail.com>

What is the socket app syntax that you are using? Do you have "async full"
turned on?
-MC

On Thu, Jun 9, 2011 at 1:22 PM, Hector Geraldino <
Hector.Geraldino at ip-soft.net> wrote:

> Hi Michael,
>
>
>
> Thanks for replying back. I?ve pasted a sample application on pastebin, the
> URL is:
>
>
>
> http://pastebin.com/MYusWe3V
>
>
>
> What I?m trying to do (and have partially accomplished) is:
>
>
>
> -Answer the call
>
> - Play a message
>
> - wait for a command (dtmf or voice, this is out of the sample app for the
> sake of keeping it short & simple)
>
> - originate a new call, wait for it to be answered and depending on an
> event (dtmf)?
>
> - ?do the bridge or dial another extension.
>
>
>
> The extension number is external (managed by a Cisco CCM). If I try to do
> the bridge from the XML dialplan it works perfect. When I try to do it from
> an outbound socket java application, anything happens, and almost no
> information is logged on FS log that I can use to decide what might be
> happening.
>
>
>
> Thanks again for your help. I truly appreciate it.
>
>
>
> Hector
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, June 09, 2011 12:47 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] originating a call from an outbound
> socket app
>
>
>
> I would say that you want to bridge the inbound leg to the outbound leg and
> use the answer confirmation variables. I recommend you pastebin a simple
> java app that answers the call and attempts to bridge it and also pastebin
> the console log. The community will take a look and see if we can help you
> figure out what is happening.
>
>
>
> -MC
>
> On Wed, Jun 8, 2011 at 3:14 PM, Hector Geraldino <
> Hector.Geraldino at ip-soft.net> wrote:
>
> Hi all,
>
>
>
> I?ve been playing with FS around weeks, with some level of success. Right
> now I?m stuck trying to dial an extension, or to bridge a call to an
> external extension. Briefly, the scenario is the following:
>
>
>
> -          Cisco CCM forwarding SIP calls to FS
>
> -          FS in outbound mode,  forwarding calls to a Java application
> using freeswitch-esl jar
>
>
>
> The java application answers the call, says a dialogue using TTS and
> receives input using DTFM or speech commands; all this is working pretty
> well. Now I?m trying to put this call in hold, originate a call to another
> extension and, depending on the result (call answered, not answered,
> destination user presses an specific number generating an expected DTMF), I
> need to bridge both calls and finish the session.
>
>
>
> I try several ways to get this done with no luck yet: using api originate
> to make the call with different arguments ({ignore_early_media,  etc). I
> even tried to skip this step and just do the bridge between the inbound call
> and the external extension, but this  didn?t work either. I can?t see any
> errors thrown by FS, and the result of the command is +OK, but nothing
> happens, the external phone never rings. The FS log only says:
>
>
>
> [DEBUG] switch_core_session.c:954 Send signal sofia/external/
> 5655 at 192.168.8.1 [BREAK] <= this ext/IP address is from the inbound
> caller, not the one I?m trying to connect to.
>
>
>
> My first impression was that I might have something wrong with the
> originate/bridge parameters. I changed the dialplan XML file to just do the
> bridge, and it worked like a charm.
>
>
>
> <include>
>
> <extension name="test">
>
>         <condition field="destination_number" expression="6114|5121">
>
> <!--    <action application="socket" data="localhost:8044 async full"/>
> -->
>
>           <action application="bridge" data="sofia/external/
> 5562 at 192.168.2.14" />
>
>         </condition>
>
> </extension>
>
> </include>
>
>
>
> I don?t know what path to follow here. I was unable to find relevant
> examples on google that guides me to solve this little issue. Any advices?
>
>
>
> Thanks all,
>
> Hector
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri Jun 10 02:12:36 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 9 Jun 2011 15:12:36 -0700
Subject: [Freeswitch-users] originating a call from an outbound socket
	app
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD021F45EB71@NY1-EXMB-01.ip-soft.net>
References: <6A6B4C284AD15042B429EB9D904544AD021F45EAA0@NY1-EXMB-01.ip-soft.net>
	<BANLkTi=LX=cGqSVrT6o-0OOPXPOA2U=MnA@mail.gmail.com>
	<6A6B4C284AD15042B429EB9D904544AD021F45EB71@NY1-EXMB-01.ip-soft.net>
Message-ID: <BANLkTikG8TQ+1JhAVr8Y6xMWhH1xqor8zA@mail.gmail.com>

Also, I don't think this is helping you:
  EslMessage response = sendSyncSingleLineCommand(channel, "myevents");

Look here to see what "myevents" does:
http://wiki.freeswitch.org/wiki/Mod_event_socket#Special_Case_-_.27myevents.27

-MC

On Thu, Jun 9, 2011 at 1:22 PM, Hector Geraldino <
Hector.Geraldino at ip-soft.net> wrote:

> Hi Michael,
>
>
>
> Thanks for replying back. I?ve pasted a sample application on pastebin, the
> URL is:
>
>
>
> http://pastebin.com/MYusWe3V
>
>
>
> What I?m trying to do (and have partially accomplished) is:
>
>
>
> -Answer the call
>
> - Play a message
>
> - wait for a command (dtmf or voice, this is out of the sample app for the
> sake of keeping it short & simple)
>
> - originate a new call, wait for it to be answered and depending on an
> event (dtmf)?
>
> - ?do the bridge or dial another extension.
>
>
>
> The extension number is external (managed by a Cisco CCM). If I try to do
> the bridge from the XML dialplan it works perfect. When I try to do it from
> an outbound socket java application, anything happens, and almost no
> information is logged on FS log that I can use to decide what might be
> happening.
>
>
>
> Thanks again for your help. I truly appreciate it.
>
>
>
> Hector
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, June 09, 2011 12:47 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] originating a call from an outbound
> socket app
>
>
>
> I would say that you want to bridge the inbound leg to the outbound leg and
> use the answer confirmation variables. I recommend you pastebin a simple
> java app that answers the call and attempts to bridge it and also pastebin
> the console log. The community will take a look and see if we can help you
> figure out what is happening.
>
>
>
> -MC
>
> On Wed, Jun 8, 2011 at 3:14 PM, Hector Geraldino <
> Hector.Geraldino at ip-soft.net> wrote:
>
> Hi all,
>
>
>
> I?ve been playing with FS around weeks, with some level of success. Right
> now I?m stuck trying to dial an extension, or to bridge a call to an
> external extension. Briefly, the scenario is the following:
>
>
>
> -          Cisco CCM forwarding SIP calls to FS
>
> -          FS in outbound mode,  forwarding calls to a Java application
> using freeswitch-esl jar
>
>
>
> The java application answers the call, says a dialogue using TTS and
> receives input using DTFM or speech commands; all this is working pretty
> well. Now I?m trying to put this call in hold, originate a call to another
> extension and, depending on the result (call answered, not answered,
> destination user presses an specific number generating an expected DTMF), I
> need to bridge both calls and finish the session.
>
>
>
> I try several ways to get this done with no luck yet: using api originate
> to make the call with different arguments ({ignore_early_media,  etc). I
> even tried to skip this step and just do the bridge between the inbound call
> and the external extension, but this  didn?t work either. I can?t see any
> errors thrown by FS, and the result of the command is +OK, but nothing
> happens, the external phone never rings. The FS log only says:
>
>
>
> [DEBUG] switch_core_session.c:954 Send signal sofia/external/
> 5655 at 192.168.8.1 [BREAK] <= this ext/IP address is from the inbound
> caller, not the one I?m trying to connect to.
>
>
>
> My first impression was that I might have something wrong with the
> originate/bridge parameters. I changed the dialplan XML file to just do the
> bridge, and it worked like a charm.
>
>
>
> <include>
>
> <extension name="test">
>
>         <condition field="destination_number" expression="6114|5121">
>
> <!--    <action application="socket" data="localhost:8044 async full"/>
> -->
>
>           <action application="bridge" data="sofia/external/
> 5562 at 192.168.2.14" />
>
>         </condition>
>
> </extension>
>
> </include>
>
>
>
> I don?t know what path to follow here. I was unable to find relevant
> examples on google that guides me to solve this little issue. Any advices?
>
>
>
> Thanks all,
>
> Hector
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From benkokakao at gmail.com  Fri Jun 10 02:52:07 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Fri, 10 Jun 2011 00:52:07 +0200
Subject: [Freeswitch-users] Voicemail Message - How t know how many are
 read & how many are unread in Inbox
In-Reply-To: <E826C94A3C6BE44198FAF4D66B5AB4D81BDA4D5BF1@VA3DIAXVS351.RED001.local>
References: <mailman.10243.1307524956.24555.freeswitch-users@lists.freeswitch.org>
	<AA1BE0FA88F3421784DBD1D91A84E515@sharad>
	<BANLkTikshbtzM67daHBePp7TB5pH5FjEyA@mail.gmail.com>
	<E826C94A3C6BE44198FAF4D66B5AB4D81BDA4D5BF1@VA3DIAXVS351.RED001.local>
Message-ID: <BANLkTikBYBK5Jugw6dtqUL88bkkic-bFAA@mail.gmail.com>

On 9 June 2011 19:43, Robert Hadley <robert.hadley at teotech.com> wrote:
>
> vm_boxcount user at domain|[new|saved|new-urgent|saved-urgent|all]]
>
> freeswitch at internal> vm_boxcount 2002 at 192.168.xx.xx|all
> 5:0:0:0

Hmm, i thought vm_boxcount is broken resp. means something different -
only displays 0 on the mailbox above(While there is 1 new and 1 saved
msg, as displayed by vm_list) :-/


From benkokakao at gmail.com  Fri Jun 10 02:55:26 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Fri, 10 Jun 2011 00:55:26 +0200
Subject: [Freeswitch-users] Enumerate variables in a session
In-Reply-To: <BANLkTikMT8GROmAdOnv8vBhuEEBYmiAeYw@mail.gmail.com>
References: <33095823FD21DF429B481B5163264B7950AC586589@VMBX102.ihostexchange.net>
	<BANLkTikmrXscsfdPUHhsgewYeF3RYNcyRQ@mail.gmail.com>
	<BANLkTikMT8GROmAdOnv8vBhuEEBYmiAeYw@mail.gmail.com>
Message-ID: <BANLkTi=moW62TNwf4q2LuubmWkJDpkTiWg@mail.gmail.com>

On 9 June 2011 16:20, Steven Ayre <steveayre at gmail.com> wrote:
> Actually Christian, it is since you have a session object in javascript and
> have to access the variables via the Session api.

Oh, sorry for that hasty reply then!


From kheimerl at cs.berkeley.edu  Fri Jun 10 07:29:21 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Thu, 9 Jun 2011 20:29:21 -0700
Subject: [Freeswitch-users] Sofia invite issues
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59DEF56421@cooper>
References: <BANLkTikWTD7bT+kiMdDk1VTOU2XsPU0rcQ@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DEE58301@cooper>
	<BANLkTinvK0rhcqov3SL8aCqp8UL4sJ9Y_Q@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DEF56421@cooper>
Message-ID: <BANLkTimK5Orq9=12M7qwm8x3qf40tJ-=Aw@mail.gmail.com>

I think you're right. Thanks! Removing IMSI641104278340883 from that
domain caused FS to fall back to the previous user, 1301.

Unfortunately, I can't lets the agents auth, or use different IPs, all
are coming from the same application that doesn't support auth.

I'm going to try adding each to their own unique domain, hopefully
that'll help. It might not though, considering they're all coming from
the same IP.

Do you have any other idea how to fix this?

On Thu, Jun 9, 2011 at 1:55 PM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> I'm guessing your problem might be related to the fact that you use the cidr attribute on the user's - with the same IP's for all of them. If I understand these correctly (never used it this way myself) they're supposed to be used to authenticate a user for a specific IP, instead of challenge auth, and since the same IP is provided for all of them they might overwrite eachother, and the last one will be the one it matches.
>
> Also, read more on http://wiki.freeswitch.org/wiki/Acl#Users.
>
> Try removing the cidr attribute, and let the agents auth instead, or use different IP's for the users.
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Kurtis Heimerl [kheimerl at cs.berkeley.edu]
> Skickat: den 9 juni 2011 22:21
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] Sofia invite issues
>
> Here you go: http://pastebin.freeswitch.org/16468
>
> For this packet trace I decided to have profile "C" in the UA
> directory. This causes FS to send the second invite (which should be
> A->B) as C->B, rather than B->B, as it does if that UA is not present.
> ?Neither is correct, this just makes it a little easier to grep.
>
> User 1302 first appears in that invite, I don't see any other log
> items talking about 1302 at all. It's worth noting that 1302's user
> name (IMSI641104878332498) DOES appear in the SDP packet loaded in the
> first (A->B) invite, but I'm pretty certain FS should not be using
> that information for SIP addressing. Again, if I remove user
> IMSI641104878332498 from the directory, FS just switched to returning
> with user 1301, it doesn't fix the issue.
>
> It's very likely I'm massaging some code that's not commonly used, as
> I may be making a small mistake here or there in the original SIP
> invite. These two lines:
> 2011-06-09 13:11:32.947157 [DEBUG] sofia.c:6674 IP 192.168.1.144
> Approved by acl "domains[IMSI641104878332498 at 192.168.1.144]". Access
> Granted.
> 2011-06-09 13:11:32.947157 [DEBUG] sofia.c:6803 Authenticating user
> IMSI641104878332498 at 192.168.1.144
> Are particularly curious. I'm not sure if those are a response to the
> incoming invite (which would be incorrect) or not. It should be
> approving user 1300 at 192.168.1.144, if anyone.
>
> Thanks!
>
> On Thu, Jun 9, 2011 at 12:18 AM, Peter Olsson
> <peter.olsson at visionutveckling.se> wrote:
>> Please pastebin a complete debug log, with "sofia global siptrace on", to http://pastebin.freeswitch.org.
>>
>> I'm pretty sure this is not a bug in FS, if it was, we would have lots of people complaining...
>>
>> /Peter
>>
>>
>> -----Ursprungligt meddelande-----
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Kurtis Heimerl
>> Skickat: den 8 juni 2011 23:32
>> Till: freeswitch-users at lists.freeswitch.org
>> ?mne: [Freeswitch-users] Sofia invite issues
>>
>> Hello Freeswitch-users!
>>
>> I'm currently writing a SIP middlebox that intercepts sip messages and
>> changes their username. The reasons for this are complicated and
>> somewhat outside of the scope of this discussion. However, I've come
>> upon a very strange issue: when making a phone-to-phone call across
>> freeswitch; it is connecting the wrong user. I'm fairly well convinced
>> this is a bug, but I thought I'd send the issue here and see if it's
>> anything I'm obviously doing wrong.
>>
>> Basically, FS sees a SIP message from a registered UA (call it A)
>> inviting another registered user to a call (B). This is acked
>> correctly (a TRYING message). FS then responds by inviting B into a
>> call with some OTHER user C. C is not mentioned at any point in the
>> initial SIP messages (as verified by wireshark). Even more strangely,
>> if I remove user C from FS (by removing their config file) FS responds
>> to the invite by inviting B into a call with itself. In each case, the
>> appropriate from header in the invite should be the original caller A.
>> ?It's worth noting that I am not monkeying with the RTP packets at
>> all, but my understanding is that SIP signalling shouldn't be affected
>> by that.
>>
>> My general guess is that I'm messing up the naming somehow, and FS is
>> running an algorithm to guess at who the call originator is. However,
>> the naming must be roughly correct; Asterisk is able to handle this
>> call just fine. I've included a sip trace of the second situation (A
>> calls B, FS invites B to a call with itself)
>>
>> For the record A:1300
>> B:1301
>> C:IMSI641104878332498
>>
>> REGISTER MESSAGE (Which works fine)
>>
>> REGISTER sip:192.168.1.144 SIP/2.0
>> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
>> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
>> From: 1300 <sip:1300 at 192.168.1.144>;tag=ftoui
>> To: 1300 <sip:1300 at 192.168.1.144>
>> Call-ID: 1032827938 at 192.168.1.144
>> CSeq: 91 REGISTER
>> Contact: <sip:1300 at 192.168.1.144:5063>;expires=7200
>> Max-Forwards: 70
>> Content-Length: 0
>>
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
>> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
>> From: 1300 <sip:1300 at 192.168.1.144>;tag=ftoui
>> To: 1300 <sip:1300 at 192.168.1.144>;tag=jU64NXypQc57F
>> Call-ID: 1032827938 at 192.168.1.144
>> CSeq: 91 REGISTER
>> Contact: <sip:1300 at 192.168.1.144:5063>;expires=7200
>> Date: Wed, 08 Jun 2011 21:02:04 GMT
>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-71d64e6 2011-06-07 22-30-16 -0700
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> Supported: timer, precondition, path, replaces
>> Content-Length: 0
>>
>> SIP TRACE (A -> B, B -> B response)
>>
>> INITIAL INVITE
>> INVITE sip:1301 at 192.168.1.144 SIP/2.0
>> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK77974
>> From: 1300 <sip:1300 at 192.168.1.144>;tag=bgdqx
>> To: <sip:1301 at 192.168.1.144>
>> Call-ID: 1817795092 at 192.168.1.144
>> CSeq: 485 INVITE
>> Contact: <sip:1300 at 192.168.1.144:5063>;expires=3600
>> Content-Type: application/sdp
>> Max-Forwards: 70
>> Content-Length: 143
>>
>> INVITE ACK
>> SIP/2.0 100 Trying
>> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK77974
>> From: 1300 <sip:1300 at 192.168.1.144>;tag=bgdqx
>> To: <sip:1301 at 192.168.1.144>
>> Call-ID: 1817795092 at 192.168.1.144
>> CSeq: 485 INVITE
>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-71d64e6 2011-06-07 22-30-16 -0700
>> Content-Length: 0
>>
>> FS INVITE (Note the from address being wrong)
>> INVITE sip:1301 at 192.168.1.144:5063 SIP/2.0
>> Via: SIP/2.0/UDP 192.168.1.144;rport;branch=z9hG4bKeeQvcyZ70SDUg
>> Max-Forwards: 69
>> From: "1301" <sip:1301 at 192.168.1.144>;tag=rHyS0Z3B61arN
>> To: <sip:1301 at 192.168.1.144:5063>
>> Call-ID: 84df47f9-0cb7-122f-13b5-5cff350d9de5
>> CSeq: 13447852 INVITE
>> Contact: <sip:mod_sofia at 192.168.1.144:5060>
>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-71d64e6 2011-06-07 22-30-16 -0700
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> Supported: timer, precondition, path, replaces
>> Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>> sla, include-session-description, presence.winfo, message-summary,
>> refer
>> Content-Type: application/sdp
>> Content-Disposition: session
>> Content-Length: 315
>> X-FS-Support: update_display
>> Remote-Party-ID: "1301"
>> <sip:1301 at 192.168.1.144>;party=calling;screen=yes;privacy=off
>>
>> USER CONFIGURATIONS
>>
>> 1300.xml
>> <include>
>> ?<user id="1300" cidr="192.168.1.0/24,127.0.0.1/32">
>> ? ?<params>
>> ? ? ?<param name="accept-blind-reg" value="true"/>
>> ? ? ?<param name="accept-blind-auth" value="true"/>
>> ? ? ?<param name="vm-password" value="1000"/>
>> ? ?</params>
>> ? ?<variables>
>> ? ? ?<variable name="toll_allow" value="domestic,international,local"/>
>> ? ? ?<variable name="accountcode" value="1300"/>
>> ? ? ?<variable name="user_context" value="default"/>
>> ? ? ?<variable name="effective_caller_id_name" value="1300"/>
>> ? ? ?<variable name="effective_caller_id_number" value="1300"/>
>> ? ? ?<variable name="outbound_caller_id_name" value="1300"/>
>> ? ? ?<variable name="outbound_caller_id_number" value="1300"/>
>> ? ? ?<variable name="callgroup" value="techsupport"/>
>> ? ?</variables>
>> ?</user>
>> </include>
>>
>> 1301.xml
>> <include>
>> ?<user id="1301" cidr="192.168.1.0/24,127.0.0.1/32">
>> ? ?<params>
>> ? ? ?<param name="accept-blind-reg" value="true"/>
>> ? ? ?<param name="accept-blind-auth" value="true"/>
>> ? ? ?<param name="vm-password" value="1000"/>
>> ? ?</params>
>> ? ?<variables>
>> ? ? ?<variable name="toll_allow" value="domestic,international,local"/>
>> ? ? ?<variable name="accountcode" value="1301"/>
>> ? ? ?<variable name="user_context" value="default"/>
>> ? ? ?<variable name="effective_caller_id_name" value="1301"/>
>> ? ? ?<variable name="effective_caller_id_number" value="1301"/>
>> ? ? ?<variable name="outbound_caller_id_name" value="1301"/>
>> ? ? ?<variable name="outbound_caller_id_number" value="1301"/>
>> ? ? ?<variable name="callgroup" value="techsupport"/>
>> ? ?</variables>
>> ?</user>
>> </include>
>>
>> IMSI641104878332498.xml
>> <include>
>> ?<user id="IMSI641104878332498" cidr="192.168.1.0/24,127.0.0.1/32"
>> number-alias="1302">
>> ? ?<params>
>> ? ? ?<param name="accept-blind-reg" value="true"/>
>> ? ? ?<param name="accept-blind-auth" value="true"/>
>> ? ? ?<param name="vm-password" value="1000"/>
>> ? ?</params>
>> ? ?<variables>
>> ? ? ?<variable name="toll_allow" value="domestic,international,local"/>
>> ? ? ?<variable name="accountcode" value="1302"/>
>> ? ? ?<variable name="user_context" value="default"/>
>> ? ? ?<variable name="effective_caller_id_name" value="IMSI641104878332498"/>
>> ? ? ?<variable name="effective_caller_id_number" value="1302"/>
>> ? ? ?<variable name="outbound_caller_id_name" value="IMSI641104878332498"/>
>> ? ? ?<variable name="outbound_caller_id_number" value="1302"/>
>> ? ? ?<variable name="callgroup" value="techsupport"/>
>> ? ?</variables>
>> ?</user>
>> </include>
>>
>> DIALPLAN
>>
>> ? ?<extension name="Local Numbers">
>> ? ? ?<condition field="destination_number" expression="^130[0-9]$">
>> ? ? ? ?<!--<action application="bridge"
>> data="user/${user_data(${destination_number}@${domain_name} attr
>> id)}"/> -->
>> ? ? ? ?<action application="bridge" data="user/${destination_number}"/>
>> ? ? ?</condition>
>> ? ?</extension>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> !DSPAM:4df12cbe32761899399581!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From OSchenk at wnr.com.au  Fri Jun 10 08:19:57 2011
From: OSchenk at wnr.com.au (Schenk, Oliver)
Date: Fri, 10 Jun 2011 12:19:57 +0800
Subject: [Freeswitch-users] CISCO 2811 - FreeSwitch - IVR
Message-ID: <A0AF161BE92F1B4CB28FDCBF2524B0420B782B@prdexch01.wnr.internal>

Hey all,

 

I have successfully managed the following setup ... at least to a
certain extent.

 

Phone line -> CISCO VIC2-4FXO -> CISCO 2811 -> Sip -> Windows Server
2008 -> FreeSwitch -> Managed IVR (IAppPlugin)

 

When I dial say the phone line's number (49794) it connects me to
extension 1024 which has the managed C#.NET IVR sitting behind it. That
works fine while everything is happy. However, when I hang up my phone.
My IVR is still happily talking to no one until the relevant timeouts
expire. Is this the normal way to implement it? Just wait for timeout?
Or do I have some CISCO configuration issues?

 

Shouldn't the CISCO tell FreeSwitch that "Hey ... the guy on the line
has put down his phone receiver so you can end your session now."?

 

 

Any comments?

 

voice-port 0/3/0

 input gain 10

 output attenuation 10

 no comfort-noise

 cptone AU

 connection plar opx 1024

 impedance complex1

!

voice-port 0/3/1

!

voice-port 0/3/2

!

voice-port 0/3/3

!

!

!

!

!

dial-peer voice 2 voip

 destination-pattern .T

 session protocol sipv2

 session target sip-server

 session transport udp

 dtmf-relay rtp-nte

 codec g711ulaw

 no vad

!

!

sip-ua

 credentials username 1000 password * ******* realm default

 authentication username 1000 password * ********

 retry invite 2

 retry register 10

 timers connect 100

 registrar ipv4:192.168.255.**:5060 expires 3600

 sip-server ipv4:192.168.255.**:5060

!

 

Regards,

 

 

Oliver Schenk

 

Control Systems Engineer

 

WestNet Rail

 


NOTICE - This e-mail and any files transmitted with it are confidential and are only for the use of the person to whom they are addressed. 
If you are not the intended recipient then you have received this e-mail in error; please advise us immediately if this is the case. 
Any views expressed in this message are those of the individual sender, except where the sender specifically states them to be the views of WestNet Rail.
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From sharad at coraltele.com  Fri Jun 10 09:17:35 2011
From: sharad at coraltele.com (sharad)
Date: Fri, 10 Jun 2011 10:47:35 +0530
Subject: [Freeswitch-users] Voicemail Message - How t know how many
	areread & how many are unread in Inbox
References: <mailman.10243.1307524956.24555.freeswitch-users@lists.freeswitch.org><AA1BE0FA88F3421784DBD1D91A84E515@sharad><BANLkTikshbtzM67daHBePp7TB5pH5FjEyA@mail.gmail.com><E826C94A3C6BE44198FAF4D66B5AB4D81BDA4D5BF1@VA3DIAXVS351.RED001.local>
	<BANLkTikBYBK5Jugw6dtqUL88bkkic-bFAA@mail.gmail.com>
Message-ID: <26DE8A8116AB4F5BB31F0675DDBFD1BF@sharad>

Thanks for your reply..

vm_boxcount is not helping me. I think I did not explain my query correctly. 
Let me explain again -

There are some messages in INBOX of a user. Some of them are read & some of 
them are new which are not read. As of now, FS marks a message as SAVED 
message when user dials 2 while listening or does not dial anything.

But in my case, neither user dialed 2 to save the msg, nor he waited for 
auto save. rather waiting, user pressed the corresponding DTMF for next 
listening message. means user has read but this mesaage is not saved.

So my query is - Is there any way to differentiate between READ / UNREAD 
messages while messages are INBOX only.

Hope it is clear now.

Thanks & regards
Sharad


----- Original Message ----- 
From: "Christian Benke" <benkokakao at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Friday, June 10, 2011 4:22 AM
Subject: Re: [Freeswitch-users] Voicemail Message - How t know how many 
areread & how many are unread in Inbox


> On 9 June 2011 19:43, Robert Hadley <robert.hadley at teotech.com> wrote:
>>
>> vm_boxcount user at domain|[new|saved|new-urgent|saved-urgent|all]]
>>
>> freeswitch at internal> vm_boxcount 2002 at 192.168.xx.xx|all
>> 5:0:0:0
>
> Hmm, i thought vm_boxcount is broken resp. means something different -
> only displays 0 on the mailbox above(While there is 1 new and 1 saved
> msg, as displayed by vm_list) :-/
>
>
> 



From u2nsam at gmail.com  Fri Jun 10 09:59:52 2011
From: u2nsam at gmail.com (Sam)
Date: Fri, 10 Jun 2011 11:29:52 +0530
Subject: [Freeswitch-users] fetch password
Message-ID: <BANLkTinbM8ZW7DCyBYWp3NoMU8fU72-wQQ@mail.gmail.com>

hello,

If i want to store the vm pasword in database , how should i go ahead with
the dialplan to fetch the pasword.

<domain name="mydomain.com"
  <user id="101">
    <params>
      <param name="vm-password" value="12345"/>
    </params>
  </user>
</domain>



Regards
San
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From benkokakao at gmail.com  Fri Jun 10 10:03:54 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Fri, 10 Jun 2011 08:03:54 +0200
Subject: [Freeswitch-users] Voicemail Message - How t know how many
 areread & how many are unread in Inbox
In-Reply-To: <26DE8A8116AB4F5BB31F0675DDBFD1BF@sharad>
References: <mailman.10243.1307524956.24555.freeswitch-users@lists.freeswitch.org>
	<AA1BE0FA88F3421784DBD1D91A84E515@sharad>
	<BANLkTikshbtzM67daHBePp7TB5pH5FjEyA@mail.gmail.com>
	<E826C94A3C6BE44198FAF4D66B5AB4D81BDA4D5BF1@VA3DIAXVS351.RED001.local>
	<BANLkTikBYBK5Jugw6dtqUL88bkkic-bFAA@mail.gmail.com>
	<26DE8A8116AB4F5BB31F0675DDBFD1BF@sharad>
Message-ID: <BANLkTiknapraGXceJXfcf9RHJXBRpbE0QQ@mail.gmail.com>

> So my query is - Is there any way to differentiate between READ / UNREAD
> messages while messages are INBOX only.

Have you checked if the "read_epoch"-parameter in the
voicemail_msgs-table in voicemail_default.db or your odbc-db is set to
a "real" value when this happens(Also displayed above with vm_list)?
You could then simply query the db (e.g. "select count(*) from
voicemail_msgs where read_epoch>0;") instead of using the commands
above.


From benkokakao at gmail.com  Fri Jun 10 10:06:39 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Fri, 10 Jun 2011 08:06:39 +0200
Subject: [Freeswitch-users] fetch password
In-Reply-To: <BANLkTinbM8ZW7DCyBYWp3NoMU8fU72-wQQ@mail.gmail.com>
References: <BANLkTinbM8ZW7DCyBYWp3NoMU8fU72-wQQ@mail.gmail.com>
Message-ID: <BANLkTimikq6z2T+thzDm8ZDDZx7yFAhUNA@mail.gmail.com>

> If i want to store the vm pasword in database , how should i go ahead with
> the dialplan to fetch the pasword.


I'm not sure if i correctly understand your request. If you want to
save the voicemail-settings in a db rather than a directory-file,
there's a schema for
that(http://wiki.freeswitch.org/wiki/Mod_voicemail#Database_Schema),
the voicemail-app then fetches the password directly from the db via
odbc. Does that help you?

Regards,
Christian


From david.ponzone at ipeva.fr  Fri Jun 10 10:32:54 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 10 Jun 2011 08:32:54 +0200
Subject: [Freeswitch-users] Default Install Extensions NEWBIE Question
In-Reply-To: <4DF0FE6F.2020806@gmail.com>
References: <4DF0FE6F.2020806@gmail.com>
Message-ID: <D52C1166-CB8F-4CB2-ACA3-9096C8CC19C4@ipeva.fr>

Pretty much, but wouldn't it be better to give it a try ?
Requirements vary...

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

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

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




Le 09/06/2011 ? 19:10, Mehma Sarja a ?crit :

> The default install on ubuntu renders a working system servicing 19 
> extensions. Can I use this as-is in a production environment which only 
> requires within-LAN, inter-departmental voice connectivity?
> 
> Mehma
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From sharad at coraltele.com  Fri Jun 10 10:38:52 2011
From: sharad at coraltele.com (sharad)
Date: Fri, 10 Jun 2011 12:08:52 +0530
Subject: [Freeswitch-users] Voicemail Message - How t know how
	manyareread & how many are unread in Inbox
References: <mailman.10243.1307524956.24555.freeswitch-users@lists.freeswitch.org><AA1BE0FA88F3421784DBD1D91A84E515@sharad><BANLkTikshbtzM67daHBePp7TB5pH5FjEyA@mail.gmail.com><E826C94A3C6BE44198FAF4D66B5AB4D81BDA4D5BF1@VA3DIAXVS351.RED001.local><BANLkTikBYBK5Jugw6dtqUL88bkkic-bFAA@mail.gmail.com><26DE8A8116AB4F5BB31F0675DDBFD1BF@sharad>
	<BANLkTiknapraGXceJXfcf9RHJXBRpbE0QQ@mail.gmail.com>
Message-ID: <DFC9276BB70D4FB99BD4FF88483B2590@sharad>

Thanks for reply,

Yes, I have checked this. The value of read_epoch is changed only if message 
is saved. If message is not saved (only listen), than it remain 0 only.

Plz advice further..

Regards
Sharad


----- Original Message ----- 
From: "Christian Benke" <benkokakao at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Friday, June 10, 2011 11:33 AM
Subject: Re: [Freeswitch-users] Voicemail Message - How t know how 
manyareread & how many are unread in Inbox


>> So my query is - Is there any way to differentiate between READ / UNREAD
>> messages while messages are INBOX only.
>
> Have you checked if the "read_epoch"-parameter in the
> voicemail_msgs-table in voicemail_default.db or your odbc-db is set to
> a "real" value when this happens(Also displayed above with vm_list)?
> You could then simply query the db (e.g. "select count(*) from
> voicemail_msgs where read_epoch>0;") instead of using the commands
> above.
>
>
> 



From peter.olsson at visionutveckling.se  Fri Jun 10 10:54:55 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 10 Jun 2011 08:54:55 +0200
Subject: [Freeswitch-users] Sofia invite issues
In-Reply-To: <BANLkTimK5Orq9=12M7qwm8x3qf40tJ-=Aw@mail.gmail.com>
References: <BANLkTikWTD7bT+kiMdDk1VTOU2XsPU0rcQ@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DEE58301@cooper>
	<BANLkTinvK0rhcqov3SL8aCqp8UL4sJ9Y_Q@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DEF56421@cooper>
	<BANLkTimK5Orq9=12M7qwm8x3qf40tJ-=Aw@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59DEE58661@cooper>

In that case, just add the cidr stuff to the global acl "domain" (acl.conf.xml under autoload_configs), this is used by default by the sofia internal profile, or create a new acl the the profile uses.

Something like;
      <node type="allow" cidr="192.168.1.0/24"/>
      <node type="allow" cidr="127.0.0.1/32"/>

This will make all calls from these IP's just to pass in. They won't be detected as any user, so the original From will be used when bridging the outgoing call.

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Kurtis Heimerl
Skickat: den 10 juni 2011 05:29
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Sofia invite issues

I think you're right. Thanks! Removing IMSI641104278340883 from that
domain caused FS to fall back to the previous user, 1301.

Unfortunately, I can't lets the agents auth, or use different IPs, all
are coming from the same application that doesn't support auth.

I'm going to try adding each to their own unique domain, hopefully
that'll help. It might not though, considering they're all coming from
the same IP.

Do you have any other idea how to fix this?

On Thu, Jun 9, 2011 at 1:55 PM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> I'm guessing your problem might be related to the fact that you use the cidr attribute on the user's - with the same IP's for all of them. If I understand these correctly (never used it this way myself) they're supposed to be used to authenticate a user for a specific IP, instead of challenge auth, and since the same IP is provided for all of them they might overwrite eachother, and the last one will be the one it matches.
>
> Also, read more on http://wiki.freeswitch.org/wiki/Acl#Users.
>
> Try removing the cidr attribute, and let the agents auth instead, or use different IP's for the users.
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Kurtis Heimerl [kheimerl at cs.berkeley.edu]
> Skickat: den 9 juni 2011 22:21
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] Sofia invite issues
>
> Here you go: http://pastebin.freeswitch.org/16468
>
> For this packet trace I decided to have profile "C" in the UA
> directory. This causes FS to send the second invite (which should be
> A->B) as C->B, rather than B->B, as it does if that UA is not present.
> ?Neither is correct, this just makes it a little easier to grep.
>
> User 1302 first appears in that invite, I don't see any other log
> items talking about 1302 at all. It's worth noting that 1302's user
> name (IMSI641104878332498) DOES appear in the SDP packet loaded in the
> first (A->B) invite, but I'm pretty certain FS should not be using
> that information for SIP addressing. Again, if I remove user
> IMSI641104878332498 from the directory, FS just switched to returning
> with user 1301, it doesn't fix the issue.
>
> It's very likely I'm massaging some code that's not commonly used, as
> I may be making a small mistake here or there in the original SIP
> invite. These two lines:
> 2011-06-09 13:11:32.947157 [DEBUG] sofia.c:6674 IP 192.168.1.144
> Approved by acl "domains[IMSI641104878332498 at 192.168.1.144]". Access
> Granted.
> 2011-06-09 13:11:32.947157 [DEBUG] sofia.c:6803 Authenticating user
> IMSI641104878332498 at 192.168.1.144
> Are particularly curious. I'm not sure if those are a response to the
> incoming invite (which would be incorrect) or not. It should be
> approving user 1300 at 192.168.1.144, if anyone.
>
> Thanks!
>
> On Thu, Jun 9, 2011 at 12:18 AM, Peter Olsson
> <peter.olsson at visionutveckling.se> wrote:
>> Please pastebin a complete debug log, with "sofia global siptrace on", to http://pastebin.freeswitch.org.
>>
>> I'm pretty sure this is not a bug in FS, if it was, we would have lots of people complaining...
>>
>> /Peter
>>
>>
>> -----Ursprungligt meddelande-----
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Kurtis Heimerl
>> Skickat: den 8 juni 2011 23:32
>> Till: freeswitch-users at lists.freeswitch.org
>> ?mne: [Freeswitch-users] Sofia invite issues
>>
>> Hello Freeswitch-users!
>>
>> I'm currently writing a SIP middlebox that intercepts sip messages and
>> changes their username. The reasons for this are complicated and
>> somewhat outside of the scope of this discussion. However, I've come
>> upon a very strange issue: when making a phone-to-phone call across
>> freeswitch; it is connecting the wrong user. I'm fairly well convinced
>> this is a bug, but I thought I'd send the issue here and see if it's
>> anything I'm obviously doing wrong.
>>
>> Basically, FS sees a SIP message from a registered UA (call it A)
>> inviting another registered user to a call (B). This is acked
>> correctly (a TRYING message). FS then responds by inviting B into a
>> call with some OTHER user C. C is not mentioned at any point in the
>> initial SIP messages (as verified by wireshark). Even more strangely,
>> if I remove user C from FS (by removing their config file) FS responds
>> to the invite by inviting B into a call with itself. In each case, the
>> appropriate from header in the invite should be the original caller A.
>> ?It's worth noting that I am not monkeying with the RTP packets at
>> all, but my understanding is that SIP signalling shouldn't be affected
>> by that.
>>
>> My general guess is that I'm messing up the naming somehow, and FS is
>> running an algorithm to guess at who the call originator is. However,
>> the naming must be roughly correct; Asterisk is able to handle this
>> call just fine. I've included a sip trace of the second situation (A
>> calls B, FS invites B to a call with itself)
>>
>> For the record A:1300
>> B:1301
>> C:IMSI641104878332498
>>
>> REGISTER MESSAGE (Which works fine)
>>
>> REGISTER sip:192.168.1.144 SIP/2.0
>> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
>> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
>> From: 1300 <sip:1300 at 192.168.1.144>;tag=ftoui
>> To: 1300 <sip:1300 at 192.168.1.144>
>> Call-ID: 1032827938 at 192.168.1.144
>> CSeq: 91 REGISTER
>> Contact: <sip:1300 at 192.168.1.144:5063>;expires=7200
>> Max-Forwards: 70
>> Content-Length: 0
>>
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
>> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
>> From: 1300 <sip:1300 at 192.168.1.144>;tag=ftoui
>> To: 1300 <sip:1300 at 192.168.1.144>;tag=jU64NXypQc57F
>> Call-ID: 1032827938 at 192.168.1.144
>> CSeq: 91 REGISTER
>> Contact: <sip:1300 at 192.168.1.144:5063>;expires=7200
>> Date: Wed, 08 Jun 2011 21:02:04 GMT
>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-71d64e6 2011-06-07 22-30-16 -0700
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> Supported: timer, precondition, path, replaces
>> Content-Length: 0
>>
>> SIP TRACE (A -> B, B -> B response)
>>
>> INITIAL INVITE
>> INVITE sip:1301 at 192.168.1.144 SIP/2.0
>> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK77974
>> From: 1300 <sip:1300 at 192.168.1.144>;tag=bgdqx
>> To: <sip:1301 at 192.168.1.144>
>> Call-ID: 1817795092 at 192.168.1.144
>> CSeq: 485 INVITE
>> Contact: <sip:1300 at 192.168.1.144:5063>;expires=3600
>> Content-Type: application/sdp
>> Max-Forwards: 70
>> Content-Length: 143
>>
>> INVITE ACK
>> SIP/2.0 100 Trying
>> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK77974
>> From: 1300 <sip:1300 at 192.168.1.144>;tag=bgdqx
>> To: <sip:1301 at 192.168.1.144>
>> Call-ID: 1817795092 at 192.168.1.144
>> CSeq: 485 INVITE
>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-71d64e6 2011-06-07 22-30-16 -0700
>> Content-Length: 0
>>
>> FS INVITE (Note the from address being wrong)
>> INVITE sip:1301 at 192.168.1.144:5063 SIP/2.0
>> Via: SIP/2.0/UDP 192.168.1.144;rport;branch=z9hG4bKeeQvcyZ70SDUg
>> Max-Forwards: 69
>> From: "1301" <sip:1301 at 192.168.1.144>;tag=rHyS0Z3B61arN
>> To: <sip:1301 at 192.168.1.144:5063>
>> Call-ID: 84df47f9-0cb7-122f-13b5-5cff350d9de5
>> CSeq: 13447852 INVITE
>> Contact: <sip:mod_sofia at 192.168.1.144:5060>
>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-71d64e6 2011-06-07 22-30-16 -0700
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> Supported: timer, precondition, path, replaces
>> Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>> sla, include-session-description, presence.winfo, message-summary,
>> refer
>> Content-Type: application/sdp
>> Content-Disposition: session
>> Content-Length: 315
>> X-FS-Support: update_display
>> Remote-Party-ID: "1301"
>> <sip:1301 at 192.168.1.144>;party=calling;screen=yes;privacy=off
>>
>> USER CONFIGURATIONS
>>
>> 1300.xml
>> <include>
>> ?<user id="1300" cidr="192.168.1.0/24,127.0.0.1/32">
>> ? ?<params>
>> ? ? ?<param name="accept-blind-reg" value="true"/>
>> ? ? ?<param name="accept-blind-auth" value="true"/>
>> ? ? ?<param name="vm-password" value="1000"/>
>> ? ?</params>
>> ? ?<variables>
>> ? ? ?<variable name="toll_allow" value="domestic,international,local"/>
>> ? ? ?<variable name="accountcode" value="1300"/>
>> ? ? ?<variable name="user_context" value="default"/>
>> ? ? ?<variable name="effective_caller_id_name" value="1300"/>
>> ? ? ?<variable name="effective_caller_id_number" value="1300"/>
>> ? ? ?<variable name="outbound_caller_id_name" value="1300"/>
>> ? ? ?<variable name="outbound_caller_id_number" value="1300"/>
>> ? ? ?<variable name="callgroup" value="techsupport"/>
>> ? ?</variables>
>> ?</user>
>> </include>
>>
>> 1301.xml
>> <include>
>> ?<user id="1301" cidr="192.168.1.0/24,127.0.0.1/32">
>> ? ?<params>
>> ? ? ?<param name="accept-blind-reg" value="true"/>
>> ? ? ?<param name="accept-blind-auth" value="true"/>
>> ? ? ?<param name="vm-password" value="1000"/>
>> ? ?</params>
>> ? ?<variables>
>> ? ? ?<variable name="toll_allow" value="domestic,international,local"/>
>> ? ? ?<variable name="accountcode" value="1301"/>
>> ? ? ?<variable name="user_context" value="default"/>
>> ? ? ?<variable name="effective_caller_id_name" value="1301"/>
>> ? ? ?<variable name="effective_caller_id_number" value="1301"/>
>> ? ? ?<variable name="outbound_caller_id_name" value="1301"/>
>> ? ? ?<variable name="outbound_caller_id_number" value="1301"/>
>> ? ? ?<variable name="callgroup" value="techsupport"/>
>> ? ?</variables>
>> ?</user>
>> </include>
>>
>> IMSI641104878332498.xml
>> <include>
>> ?<user id="IMSI641104878332498" cidr="192.168.1.0/24,127.0.0.1/32"
>> number-alias="1302">
>> ? ?<params>
>> ? ? ?<param name="accept-blind-reg" value="true"/>
>> ? ? ?<param name="accept-blind-auth" value="true"/>
>> ? ? ?<param name="vm-password" value="1000"/>
>> ? ?</params>
>> ? ?<variables>
>> ? ? ?<variable name="toll_allow" value="domestic,international,local"/>
>> ? ? ?<variable name="accountcode" value="1302"/>
>> ? ? ?<variable name="user_context" value="default"/>
>> ? ? ?<variable name="effective_caller_id_name" value="IMSI641104878332498"/>
>> ? ? ?<variable name="effective_caller_id_number" value="1302"/>
>> ? ? ?<variable name="outbound_caller_id_name" value="IMSI641104878332498"/>
>> ? ? ?<variable name="outbound_caller_id_number" value="1302"/>
>> ? ? ?<variable name="callgroup" value="techsupport"/>
>> ? ?</variables>
>> ?</user>
>> </include>
>>
>> DIALPLAN
>>
>> ? ?<extension name="Local Numbers">
>> ? ? ?<condition field="destination_number" expression="^130[0-9]$">
>> ? ? ? ?<!--<action application="bridge"
>> data="user/${user_data(${destination_number}@${domain_name} attr
>> id)}"/> -->
>> ? ? ? ?<action application="bridge" data="user/${destination_number}"/>
>> ? ? ?</condition>
>> ? ?</extension>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> 
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

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

!DSPAM:4df1916532761203114732!



From u2nsam at gmail.com  Fri Jun 10 11:38:16 2011
From: u2nsam at gmail.com (Sam)
Date: Fri, 10 Jun 2011 13:08:16 +0530
Subject: [Freeswitch-users] fetch password
In-Reply-To: <BANLkTimikq6z2T+thzDm8ZDDZx7yFAhUNA@mail.gmail.com>
References: <BANLkTinbM8ZW7DCyBYWp3NoMU8fU72-wQQ@mail.gmail.com>
	<BANLkTimikq6z2T+thzDm8ZDDZx7yFAhUNA@mail.gmail.com>
Message-ID: <BANLkTimZv5uM8+atzR15qds8Ap3UHVbU6Q@mail.gmail.com>

How Does it access database , does we need to write some syntax somewhere,
so that FS fetches it out from DB.

Regds
Sam

On Fri, Jun 10, 2011 at 11:36 AM, Christian Benke <benkokakao at gmail.com>wrote:

> > If i want to store the vm pasword in database , how should i go ahead
> with
> > the dialplan to fetch the pasword.
>
>
> I'm not sure if i correctly understand your request. If you want to
> save the voicemail-settings in a db rather than a directory-file,
> there's a schema for
> that(http://wiki.freeswitch.org/wiki/Mod_voicemail#Database_Schema),
> the voicemail-app then fetches the password directly from the db via
> odbc. Does that help you?
>
> Regards,
> Christian
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From nazim.aghabayov at gmail.com  Fri Jun 10 11:41:44 2011
From: nazim.aghabayov at gmail.com (Nazim Aghabayov)
Date: Fri, 10 Jun 2011 12:41:44 +0500
Subject: [Freeswitch-users] FreeTDM - Sangoma B700 - ISDN connection
 questions - UK
In-Reply-To: <4DED4854.6010203@earthspike.net>
References: <4DED4854.6010203@earthspike.net>
Message-ID: <4DF1CAB8.3070706@gmail.com>

Hello John,

I'm not an ISDN expert, but have some experience with PRI.
Have you checked the line? Sometimes bad physical interconnects makes 
line noisy.
Channels going up and down may indicate a physical line problem or 
inadequate grounding (or mixed signal / common grounds). I would disable 
hardware echo-canellation and hw. dtmf during the troubleshooting.
On PRI it's possible to trace the D channel with "wanpipemon -i w1g1 
-pcap -pcap_file isdn.pcap -prot ISDN -full -systime -c trd". It should 
be supported on BRI too, look through wanpipemon manual.

Best Regards,
Nazim


On 06/07/2011 02:36 AM, John wrote:
> Hello,
>
> I have just set up a FreeSWITCH box with a Sangoma B700 connected to 2 x
> ISDN2e lines (each 2B+D, so 4 voice channels) in UK, so supplied by BT
> Openreach.  There are a number of anomalies that I am trying to solve.
> [One of these is that 1 line is dead, but that is for BT Openreach to
> resolve rather than anyone on this list.]
>
> I have a few questions, but as they are all related, I hope you don't
> mind them in one post.  Some basics first.  The box is an Atom dual-core
> with 2GB of memory and a Sangoma B700 card.  It's built with Ubuntu
> 10.04.2 LTS server 64-bit, patched and up to date, and also runs dhcpd,
> lighttpd and sshd.  I built the Sangoma ISDN libraries and FreeSWITCH
> using the latest git versions I could ('make current' about 2 weeks
> ago).  We have incoming and outgoing calls working, but some incoming
> calls ring in the caller's ear, but nothing appears on the FreeSWITCH
> console, and others fail.  Rebooting the server cures this.  We have had
> problems with lines being disconnected and then reconnected, and it
> seems that FreeTDM/wanrouter/whatever doesn't recognise previously
> disconnected lines coming back into use, because a reboot finds lines
> that were previously reported disconnected ('wanrouter status' or 'ftdm
> list').  We have ongoing problems with one line that is 'disconnected'
> (wanpipe2/wp2) but the queries I am describing below apply equally when
> both lines are connected and working.
>
> 1. There seems to be a lot of ISDN 'chatter' with channels going up and
> down all the time even when the switch is completely idle.  Is this
> normal?  Here is my /log 7 showing two of the cycles (which appear to be
> about 50s apart):
>
> 2011-06-06 22:01:54.992467 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:681
> [SNGISDN Q921] wp1: Protocol: Data Link connection DN(4): Disconnect
> initiated(263)
> 2011-06-06 22:02:29.952464 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:748
> [SNGISDN Q931] s1: Interface: Down(261): Dchan(285)
> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
> [s1c1][1:1] Signalling link status changed to DOWN
> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
> [s1c2][1:2] Signalling link status changed to DOWN
> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
> [s1c3][1:3] Signalling link status changed to DOWN
> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:200
> [s1c1][1:1] Setting availability rate to:5
> 2011-06-06 22:02:29.952464 [DEBUG] mod_freetdm.c:2268 got clear channel
> sig [SIGSTATUS_CHANGED]
> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:186
> [s1c1][1:1] Setting availability rate to:5
> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:200
> [s1c2][1:2] Setting availability rate to:5
> 2011-06-06 22:02:29.952464 [INFO] mod_freetdm.c:2345 1:1 signalling
> changed to :DOWN
> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:186
> [s1c2][1:2] Setting availability rate to:5
> 2011-06-06 22:02:29.952464 [DEBUG] mod_freetdm.c:2268 got clear channel
> sig [SIGSTATUS_CHANGED]
> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:200
> [s1c3][1:3] Setting availability rate to:5
> 2011-06-06 22:02:29.952464 [INFO] mod_freetdm.c:2345 1:2 signalling
> changed to :DOWN
> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:186
> [s1c3][1:3] Setting availability rate to:5
> 2011-06-06 22:02:29.952464 [DEBUG] mod_freetdm.c:2268 got clear channel
> sig [SIGSTATUS_CHANGED]
> 2011-06-06 22:02:29.952464 [INFO] mod_freetdm.c:2345 1:3 signalling
> changed to :DOWN
> 2011-06-06 22:02:29.952464 [INFO] ftmod_sangoma_isdn_stack_rcv.c:631
> Received RESTART CFM (dChan:1 ces:0 type:1)
> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:1179
> Receved RESTART, but Restart Indicator IE not present
> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:681
> [SNGISDN Q921] wp1: Protocol: Data Link connection UP(3): UA frame with
> F-bit = 1(258)
> 2011-06-06 22:02:29.972464 [INFO] ftmod_sangoma_isdn_stack_rcv.c:631
> Received RESTART CFM (dChan:1 ces:0 type:0)
> 2011-06-06 22:02:29.972464 [INFO] ftmod_sangoma_isdn_stack_rcv.c:729
> [SNGISDN Q931] s1: Interface: UP(260): Dchan(285)
> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
> [s1c1][1:1] Signalling link status changed to UP
> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
> [s1c2][1:2] Signalling link status changed to UP
> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
> [s1c3][1:3] Signalling link status changed to UP
> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:200
> [s1c1][1:1] Setting availability rate to:10
> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:186
> [s1c1][1:1] Setting availability rate to:10
> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:200
> [s1c2][1:2] Setting availability rate to:10
> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:186
> [s1c2][1:2] Setting availability rate to:10
> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:200
> [s1c3][1:3] Setting availability rate to:10
> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:186
> [s1c3][1:3] Setting availability rate to:10
> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:1179
> Receved RESTART, but Restart Indicator IE not present
> 2011-06-06 22:02:29.972464 [DEBUG] mod_freetdm.c:2268 got clear channel
> sig [SIGSTATUS_CHANGED]
> 2011-06-06 22:02:29.972464 [INFO] mod_freetdm.c:2345 1:1 signalling
> changed to :UP
> 2011-06-06 22:02:29.972464 [DEBUG] mod_freetdm.c:2268 got clear channel
> sig [SIGSTATUS_CHANGED]
> 2011-06-06 22:02:29.972464 [INFO] mod_freetdm.c:2345 1:2 signalling
> changed to :UP
> 2011-06-06 22:02:29.972464 [DEBUG] mod_freetdm.c:2268 got clear channel
> sig [SIGSTATUS_CHANGED]
> 2011-06-06 22:02:29.972464 [INFO] mod_freetdm.c:2345 1:3 signalling
> changed to :UP
> 2011-06-06 22:03:19.912462 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:681
> [SNGISDN Q921] wp1: Protocol: Data Link connection DN(4): Disconnect
> initiated(263)
> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:681
> [SNGISDN Q921] wp1: Protocol: Data Link connection UP(3): UA frame with
> F-bit = 1(258)
> 2011-06-06 22:03:19.932483 [INFO] ftmod_sangoma_isdn_stack_rcv.c:631
> Received RESTART CFM (dChan:1 ces:0 type:0)
> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:1179
> Receved RESTART, but Restart Indicator IE not present
> 2011-06-06 22:03:19.932483 [INFO] ftmod_sangoma_isdn_stack_rcv.c:729
> [SNGISDN Q931] s1: Interface: UP(260): Dchan(285)
> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
> [s1c1][1:1] Signalling link status changed to UP
> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
> [s1c2][1:2] Signalling link status changed to UP
> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
> [s1c3][1:3] Signalling link status changed to UP
> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:200
> [s1c1][1:1] Setting availability rate to:10
> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:186
> [s1c1][1:1] Setting availability rate to:10
> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:200
> [s1c2][1:2] Setting availability rate to:10
> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:186
> [s1c2][1:2] Setting availability rate to:10
> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:200
> [s1c3][1:3] Setting availability rate to:10
> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:186
> [s1c3][1:3] Setting availability rate to:10
> 2011-06-06 22:03:19.932483 [DEBUG] mod_freetdm.c:2268 got clear channel
> sig [SIGSTATUS_CHANGED]
> 2011-06-06 22:03:19.932483 [INFO] mod_freetdm.c:2345 1:1 signalling
> changed to :UP
> 2011-06-06 22:03:19.932483 [DEBUG] mod_freetdm.c:2268 got clear channel
> sig [SIGSTATUS_CHANGED]
> 2011-06-06 22:03:19.932483 [INFO] mod_freetdm.c:2345 1:2 signalling
> changed to :UP
> 2011-06-06 22:03:19.932483 [DEBUG] mod_freetdm.c:2268 got clear channel
> sig [SIGSTATUS_CHANGED]
> 2011-06-06 22:03:19.932483 [INFO] mod_freetdm.c:2345 1:3 signalling
> changed to :UP
> freeswitch at internal>  ftdm list
> +OK
> span: 1 (wp1)
> type: Sangoma (ISDN)
> physical_status: ok
> signaling_status: UP
> chan_count: 3
> dialplan: XML
> context: public
> dial_regex:
> fail_dial_regex:
> hold_music:
> analog_options: none
> +OK
> span: 2 (wp2)
> type: Sangoma (ISDN)
> physical_status: alarmed
> signaling_status: DOWN
> chan_count: 3
> dialplan: XML
> context: public
> dial_regex:
> fail_dial_regex:
> hold_music:
> analog_options: none
> +OK
> span: 3 (FXS)
> type: analog
> physical_status: ok
> signaling_status: UP
> chan_count: 2
> dialplan: XML
> context: default
> dial_regex:
> fail_dial_regex:
> hold_music:
> analog_options: none
>
>
> # wanrouter status
>
> Devices currently active:
>       wanpipe1 wanpipe2 wanpipe3
>
>
> Wanpipe Config:
>
> Device name | Protocol Map | Adapter  | IRQ | Slot/IO | If's | CLK |
> Baud rate |
> wanpipe1    | N/A          | A500/B700| 20  | 0       | 1    | N/A |
> 0         |
> wanpipe2    | N/A          | A500/B700| 20  | 0       | 1    | N/A |
> 0         |
> wanpipe3    | N/A          | A200/A400/B600/B700/B800| 20  | 0       |
> 1    | N/A | 0         |
>
> Wanrouter Status:
>
> Device name | Protocol | Station | Status        |
> wanpipe1    | AFT ISDN | N/A     | Connected     |
> wanpipe2    | AFT ISDN | N/A     | Disconnected  |
> wanpipe3    | A-ANALOG | N/A     | Connected     |
>
>
> # cat /etc/wanpipe/wanrouter.rc
> #!/bin/sh
> # .. comments snipped ...
> ROUTER_BOOT=YES
> WAN_CONF_DIR=/etc/wanpipe
> WAN_INTR_DIR=/etc/wanpipe/interfaces
> WAN_BIN_DIR=/usr/sbin
> WAN_LOG=/var/log/wanrouter
> WAN_LOCK=/var/lock/wanrouter
> WAN_LOCK_DIR=/var/lock
> WAN_IP_FORWARD=NO
> NEW_IF_TYPE=NO
> WAN_LIB_DIR=/etc/wanpipe/lib
> WAN_ADSL_LIST=/etc/wanpipe/wan_adsl.list
> WAN_ANNEXG_LOAD=NO
> WAN_SCTP_LOAD=NO
> WAN_LIP_LOAD=NO
> WAN_DYN_WANCONFIG=NO
> WAN_SCRIPTS_DIR=/etc/wanpipe/scripts
> WAN_FIRMWARE_DIR=/etc/wanpipe/firmware
> WAN_DEVICES_REV_STOP_ORDER=YES
> WAN_DEVICES="wanpipe1 wanpipe2 wanpipe3 "
>
> # cat /etc/wanpipe/wanpipe1.conf
> #================================================
> # WANPIPE1 Configuration File
> #================================================
> #
> # Note: This file was generated automatically
> #       by /usr/local/sbin/setup-sangoma program.
> #
> #       If you want to edit this file, it is
> #       recommended that you use wancfg program
> #       to do so.
> #================================================
> # Sangoma Technologies Inc.
> #================================================
>
>
>
>
> [devices]
> wanpipe1 = WAN_AFT_ISDN_BRI, Comment
>
> [interfaces]
> w1g1 = wanpipe1, , TDM_VOICE_API, Comment
>
> [wanpipe1]
> CARD_TYPE     = AFT
> S514CPU     = A
> CommPort     = PRI
> AUTO_PCISLOT     = NO
> PCISLOT     = 0
> PCIBUS      = 5
> FE_MEDIA    = BRI
> FE_LINE        = 1
> TDMV_LAW    = ALAW
> RM_BRI_CLOCK_MASTER    = NO
> MTU         = 1500
> UDPPORT     = 9000
> TTL        = 255
> IGNORE_FRONT_END = NO
> TDMV_SPAN    = 1
> TE_AIS_MAINTENANCE = NO            #NO: defualt  YES: Start port in AIS
> Blue Alarm and keep line down
>                                   #wanpipemon -i w1g1 -c Ttx_ais_off to
> disable AIS maintenance mode
>                                   #wanpipemon -i w1g1 -c Ttx_ais_on to
> enable AIS maintenance mode
> TDMV_HW_DTMF        = YES        # YES: receive dtmf events from hardware
> TDMV_HW_FAX_DETECT      = YES     # YES: receive fax 1100hz events from
> hardware
> HWEC_OPERATION_MODE     = OCT_NORMAL    # OCT_NORMAL: echo cancelation
> enabled with nlp (default)
>                                           # OCT_SPEECH: improves software
> tone detection by disabling NLP (echo possible)
>                                           # OCT_NO_ECHO:disables echo
> cancelation but allows VQE/tone functions.
> HWEC_DTMF_REMOVAL       = NO    # NO: default  YES: remove dtmf out of
> incoming media (must have hwdtmf enabled)
> HWEC_NOISE_REDUCTION    = NO    # NO: default  YES: reduces noise on the
> line - could break fax
> HWEC_ACUSTIC_ECHO       = NO    # NO: default  YES: enables acustic echo
> cancelation
> HWEC_NLP_DISABLE        = NO    # NO: default  YES: guarantees software
> tone detection (possible echo)
> HWEC_TX_AUTO_GAIN       = 0     # 0: disable   -40-0: default tx audio
> level to be maintained (-20 default)
> HWEC_RX_AUTO_GAIN       = 0     # 0: disable   -40-0: default tx audio
> level to be maintained (-20 default)
> HWEC_TX_GAIN            = 0     # 0: disable   -24-24: db values to be
> applied to tx signal
> HWEC_RX_GAIN            = 0     # 0: disable   -24-24: db values to be
> applied to tx signal
>
> [w1g1]
> ACTIVE_CH    = ALL
> TDMV_HWEC    = YES
> MTU         = 80
>
>    ... and wanpipe2 was also automatically generated so just has '2' for
> FE_LINE and TDMV_SPAN instead of '1'.
>
> # cat /usr/local/freeswitch/conf/freetdm.conf
> [span wanpipe wp1]
> trunk_type =>  bri
> group=1
> b-channel =>  1:1-2
> d-channel =>  1:3
>
> [span wanpipe wp2]
> trunk_type =>  bri
> group=1
> b-channel =>  2:1-2
> d-channel =>  2:3
>
> [span wanpipe FXS]
> name =>  freetdm
> trunk_type =>  fxs
> group =>  grp2
> fxs-channel =>  3:1
>
> trunk_type =>  fxs
> group =>  grp2
> fxs-channel =>  3:2
>
> # cat /usr/local/freeswitch/conf/autoload_configs/freetdm.conf.xml
> <configuration name="freetdm.conf" description="Freetdm Configuration">
> <settings>
> <param name="debug" value="0"/>
> <!--<param name="hold-music" value="$${moh_uri}"/>-->
> <!--<param name="enable-analog-option" value="call-swap"/>-->
> <!--<param name="enable-analog-option" value="3-way"/>-->
> </settings>
>
> <config_profiles>
>
> <profile name="my_bri_te_1">
> <param name="switchtype" value="euroisdn" />
> <param name="interface" value="cpe"/>
> <param name="facility" value="no" />
> </profile>
>
> </config_profiles>
>
> <sangoma_bri_spans>
>
> <span name="wp1" cfgprofile="my_bri_te_1">
> <param name="dialplan" value="XML"/>
> <param name="context" value="public"/>
> </span>
>
> <span name="wp2" cfgprofile="my_bri_te_1">
> <param name="dialplan" value="XML"/>
> <param name="context" value="public"/>
> </span>
>
> </sangoma_bri_spans>
>
>
> <analog_spans>
>
> <span name="FXS">
> <!--<param name="hold-music" value="$${moh_uri}"/>-->
> <param name="dialplan" value="XML"/>
> <param name="context" value="default"/>
> <!-- regex to stop dialing when it matches -->
> <!--<param name="dial-regex" value="5555"/>-->
> <!-- regex to stop dialing when it does not match -->
> <!--<param name="fail-dial-regex" value="^5"/>-->
> </span>
>
> </analog_spans>
>
> </configuration>
>
> What other config files do I need to supply? I don't want to deluge the
> list any more in my first post!
>
> 2. We have 2 Single Number DDI numbers configured over the 4 channels.
> I want to set up outgoing calls so that they can appear to come from
> either of the two numbers, but at present the outgoing CLI appears to be
> overridden by the telco (BT) to only use one of the numbers.  Has anyone
> got this working in UK, and what is the format for the
> outbound_caller_id_number: last 6 digits or full 11 digits?  I note that
> the inbound called number is only the last 6 digits.
>
> 3. I have built FreeSWITCH from git and installed at /usr/local/... and
> then followed the steps on the Ubuntu page in the Wiki to set up the run
> control scripts, etc, and run FS non-root as freeswitch:daemon.  With
> FreeTDM, I have discovered that the /dev/wan* devices are owned by
> root:root, and so are inaccessible to FS running as non-root.  So for
> now I have added a line in /etc/init.d/freeswitch to 'chgrp freeswitch
> /dev/wan*'.  This is not the most elegant solution, because 'wanrouter
> restart' (which seems to be my best friend at the moment) resets the
> ownership to root:root.  I have tried grepping to see where the mknods
> are for these devices, but have been unsuccessful.  Is there a better
> place to 'permanently' change the device ownership?
>
> Thanks for all the great support I have already got just from editing my
> Wiki User page; this is a friendly group!
>
> John
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From freeswitch at earthspike.net  Fri Jun 10 12:01:42 2011
From: freeswitch at earthspike.net (John)
Date: Fri, 10 Jun 2011 09:01:42 +0100
Subject: [Freeswitch-users] FreeTDM - Sangoma B700 - ISDN connection
 questions - UK
In-Reply-To: <4DF1CAB8.3070706@gmail.com>
References: <4DED4854.6010203@earthspike.net> <4DF1CAB8.3070706@gmail.com>
Message-ID: <4DF1CF66.4050700@earthspike.net>

Nazim,

Late last night this got resolved; sorry I didn't post then.  We have 
had problems with the second line, and when these were fixed last night, 
the problem disappeared.  So now I have a nice stable FreeSWITCH with 
the only outputs on /log 7 being SIP re-registrations.  So, you are 
probably correct, and it was a line quality issue.

I've also resolved the device permissions issue with help from Sangoma 
support, and will post on that/update the Ubuntu Wiki page when I get a 
moment, probably after the weekend.

Thanks,

John

On 10/06/11 08:41, Nazim Aghabayov wrote:
> Hello John,
>
> I'm not an ISDN expert, but have some experience with PRI.
> Have you checked the line? Sometimes bad physical interconnects makes
> line noisy.
> Channels going up and down may indicate a physical line problem or
> inadequate grounding (or mixed signal / common grounds). I would disable
> hardware echo-canellation and hw. dtmf during the troubleshooting.
> On PRI it's possible to trace the D channel with "wanpipemon -i w1g1
> -pcap -pcap_file isdn.pcap -prot ISDN -full -systime -c trd". It should
> be supported on BRI too, look through wanpipemon manual.
>
> Best Regards,
> Nazim
>
>
> On 06/07/2011 02:36 AM, John wrote:
>> Hello,
>>
>> I have just set up a FreeSWITCH box with a Sangoma B700 connected to 2 x
>> ISDN2e lines (each 2B+D, so 4 voice channels) in UK, so supplied by BT
>> Openreach.  There are a number of anomalies that I am trying to solve.
>> [One of these is that 1 line is dead, but that is for BT Openreach to
>> resolve rather than anyone on this list.]
>>
>> I have a few questions, but as they are all related, I hope you don't
>> mind them in one post.  Some basics first.  The box is an Atom dual-core
>> with 2GB of memory and a Sangoma B700 card.  It's built with Ubuntu
>> 10.04.2 LTS server 64-bit, patched and up to date, and also runs dhcpd,
>> lighttpd and sshd.  I built the Sangoma ISDN libraries and FreeSWITCH
>> using the latest git versions I could ('make current' about 2 weeks
>> ago).  We have incoming and outgoing calls working, but some incoming
>> calls ring in the caller's ear, but nothing appears on the FreeSWITCH
>> console, and others fail.  Rebooting the server cures this.  We have had
>> problems with lines being disconnected and then reconnected, and it
>> seems that FreeTDM/wanrouter/whatever doesn't recognise previously
>> disconnected lines coming back into use, because a reboot finds lines
>> that were previously reported disconnected ('wanrouter status' or 'ftdm
>> list').  We have ongoing problems with one line that is 'disconnected'
>> (wanpipe2/wp2) but the queries I am describing below apply equally when
>> both lines are connected and working.
>>
>> 1. There seems to be a lot of ISDN 'chatter' with channels going up and
>> down all the time even when the switch is completely idle.  Is this
>> normal?  Here is my /log 7 showing two of the cycles (which appear to be
>> about 50s apart):
>>
>> 2011-06-06 22:01:54.992467 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:681
>> [SNGISDN Q921] wp1: Protocol: Data Link connection DN(4): Disconnect
>> initiated(263)
>> 2011-06-06 22:02:29.952464 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:748
>> [SNGISDN Q931] s1: Interface: Down(261): Dchan(285)
>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
>> [s1c1][1:1] Signalling link status changed to DOWN
>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
>> [s1c2][1:2] Signalling link status changed to DOWN
>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
>> [s1c3][1:3] Signalling link status changed to DOWN
>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:200
>> [s1c1][1:1] Setting availability rate to:5
>> 2011-06-06 22:02:29.952464 [DEBUG] mod_freetdm.c:2268 got clear channel
>> sig [SIGSTATUS_CHANGED]
>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:186
>> [s1c1][1:1] Setting availability rate to:5
>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:200
>> [s1c2][1:2] Setting availability rate to:5
>> 2011-06-06 22:02:29.952464 [INFO] mod_freetdm.c:2345 1:1 signalling
>> changed to :DOWN
>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:186
>> [s1c2][1:2] Setting availability rate to:5
>> 2011-06-06 22:02:29.952464 [DEBUG] mod_freetdm.c:2268 got clear channel
>> sig [SIGSTATUS_CHANGED]
>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:200
>> [s1c3][1:3] Setting availability rate to:5
>> 2011-06-06 22:02:29.952464 [INFO] mod_freetdm.c:2345 1:2 signalling
>> changed to :DOWN
>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:186
>> [s1c3][1:3] Setting availability rate to:5
>> 2011-06-06 22:02:29.952464 [DEBUG] mod_freetdm.c:2268 got clear channel
>> sig [SIGSTATUS_CHANGED]
>> 2011-06-06 22:02:29.952464 [INFO] mod_freetdm.c:2345 1:3 signalling
>> changed to :DOWN
>> 2011-06-06 22:02:29.952464 [INFO] ftmod_sangoma_isdn_stack_rcv.c:631
>> Received RESTART CFM (dChan:1 ces:0 type:1)
>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:1179
>> Receved RESTART, but Restart Indicator IE not present
>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:681
>> [SNGISDN Q921] wp1: Protocol: Data Link connection UP(3): UA frame with
>> F-bit = 1(258)
>> 2011-06-06 22:02:29.972464 [INFO] ftmod_sangoma_isdn_stack_rcv.c:631
>> Received RESTART CFM (dChan:1 ces:0 type:0)
>> 2011-06-06 22:02:29.972464 [INFO] ftmod_sangoma_isdn_stack_rcv.c:729
>> [SNGISDN Q931] s1: Interface: UP(260): Dchan(285)
>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
>> [s1c1][1:1] Signalling link status changed to UP
>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
>> [s1c2][1:2] Signalling link status changed to UP
>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
>> [s1c3][1:3] Signalling link status changed to UP
>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:200
>> [s1c1][1:1] Setting availability rate to:10
>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:186
>> [s1c1][1:1] Setting availability rate to:10
>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:200
>> [s1c2][1:2] Setting availability rate to:10
>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:186
>> [s1c2][1:2] Setting availability rate to:10
>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:200
>> [s1c3][1:3] Setting availability rate to:10
>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:186
>> [s1c3][1:3] Setting availability rate to:10
>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:1179
>> Receved RESTART, but Restart Indicator IE not present
>> 2011-06-06 22:02:29.972464 [DEBUG] mod_freetdm.c:2268 got clear channel
>> sig [SIGSTATUS_CHANGED]
>> 2011-06-06 22:02:29.972464 [INFO] mod_freetdm.c:2345 1:1 signalling
>> changed to :UP
>> 2011-06-06 22:02:29.972464 [DEBUG] mod_freetdm.c:2268 got clear channel
>> sig [SIGSTATUS_CHANGED]
>> 2011-06-06 22:02:29.972464 [INFO] mod_freetdm.c:2345 1:2 signalling
>> changed to :UP
>> 2011-06-06 22:02:29.972464 [DEBUG] mod_freetdm.c:2268 got clear channel
>> sig [SIGSTATUS_CHANGED]
>> 2011-06-06 22:02:29.972464 [INFO] mod_freetdm.c:2345 1:3 signalling
>> changed to :UP
>> 2011-06-06 22:03:19.912462 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:681
>> [SNGISDN Q921] wp1: Protocol: Data Link connection DN(4): Disconnect
>> initiated(263)
>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:681
>> [SNGISDN Q921] wp1: Protocol: Data Link connection UP(3): UA frame with
>> F-bit = 1(258)
>> 2011-06-06 22:03:19.932483 [INFO] ftmod_sangoma_isdn_stack_rcv.c:631
>> Received RESTART CFM (dChan:1 ces:0 type:0)
>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:1179
>> Receved RESTART, but Restart Indicator IE not present
>> 2011-06-06 22:03:19.932483 [INFO] ftmod_sangoma_isdn_stack_rcv.c:729
>> [SNGISDN Q931] s1: Interface: UP(260): Dchan(285)
>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
>> [s1c1][1:1] Signalling link status changed to UP
>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
>> [s1c2][1:2] Signalling link status changed to UP
>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
>> [s1c3][1:3] Signalling link status changed to UP
>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:200
>> [s1c1][1:1] Setting availability rate to:10
>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:186
>> [s1c1][1:1] Setting availability rate to:10
>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:200
>> [s1c2][1:2] Setting availability rate to:10
>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:186
>> [s1c2][1:2] Setting availability rate to:10
>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:200
>> [s1c3][1:3] Setting availability rate to:10
>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:186
>> [s1c3][1:3] Setting availability rate to:10
>> 2011-06-06 22:03:19.932483 [DEBUG] mod_freetdm.c:2268 got clear channel
>> sig [SIGSTATUS_CHANGED]
>> 2011-06-06 22:03:19.932483 [INFO] mod_freetdm.c:2345 1:1 signalling
>> changed to :UP
>> 2011-06-06 22:03:19.932483 [DEBUG] mod_freetdm.c:2268 got clear channel
>> sig [SIGSTATUS_CHANGED]
>> 2011-06-06 22:03:19.932483 [INFO] mod_freetdm.c:2345 1:2 signalling
>> changed to :UP
>> 2011-06-06 22:03:19.932483 [DEBUG] mod_freetdm.c:2268 got clear channel
>> sig [SIGSTATUS_CHANGED]
>> 2011-06-06 22:03:19.932483 [INFO] mod_freetdm.c:2345 1:3 signalling
>> changed to :UP
>> freeswitch at internal>   ftdm list
>> +OK
>> span: 1 (wp1)
>> type: Sangoma (ISDN)
>> physical_status: ok
>> signaling_status: UP
>> chan_count: 3
>> dialplan: XML
>> context: public
>> dial_regex:
>> fail_dial_regex:
>> hold_music:
>> analog_options: none
>> +OK
>> span: 2 (wp2)
>> type: Sangoma (ISDN)
>> physical_status: alarmed
>> signaling_status: DOWN
>> chan_count: 3
>> dialplan: XML
>> context: public
>> dial_regex:
>> fail_dial_regex:
>> hold_music:
>> analog_options: none
>> +OK
>> span: 3 (FXS)
>> type: analog
>> physical_status: ok
>> signaling_status: UP
>> chan_count: 2
>> dialplan: XML
>> context: default
>> dial_regex:
>> fail_dial_regex:
>> hold_music:
>> analog_options: none
>>
>>
>> # wanrouter status
>>
>> Devices currently active:
>>        wanpipe1 wanpipe2 wanpipe3
>>
>>
>> Wanpipe Config:
>>
>> Device name | Protocol Map | Adapter  | IRQ | Slot/IO | If's | CLK |
>> Baud rate |
>> wanpipe1    | N/A          | A500/B700| 20  | 0       | 1    | N/A |
>> 0         |
>> wanpipe2    | N/A          | A500/B700| 20  | 0       | 1    | N/A |
>> 0         |
>> wanpipe3    | N/A          | A200/A400/B600/B700/B800| 20  | 0       |
>> 1    | N/A | 0         |
>>
>> Wanrouter Status:
>>
>> Device name | Protocol | Station | Status        |
>> wanpipe1    | AFT ISDN | N/A     | Connected     |
>> wanpipe2    | AFT ISDN | N/A     | Disconnected  |
>> wanpipe3    | A-ANALOG | N/A     | Connected     |
>>
>>
>> # cat /etc/wanpipe/wanrouter.rc
>> #!/bin/sh
>> # .. comments snipped ...
>> ROUTER_BOOT=YES
>> WAN_CONF_DIR=/etc/wanpipe
>> WAN_INTR_DIR=/etc/wanpipe/interfaces
>> WAN_BIN_DIR=/usr/sbin
>> WAN_LOG=/var/log/wanrouter
>> WAN_LOCK=/var/lock/wanrouter
>> WAN_LOCK_DIR=/var/lock
>> WAN_IP_FORWARD=NO
>> NEW_IF_TYPE=NO
>> WAN_LIB_DIR=/etc/wanpipe/lib
>> WAN_ADSL_LIST=/etc/wanpipe/wan_adsl.list
>> WAN_ANNEXG_LOAD=NO
>> WAN_SCTP_LOAD=NO
>> WAN_LIP_LOAD=NO
>> WAN_DYN_WANCONFIG=NO
>> WAN_SCRIPTS_DIR=/etc/wanpipe/scripts
>> WAN_FIRMWARE_DIR=/etc/wanpipe/firmware
>> WAN_DEVICES_REV_STOP_ORDER=YES
>> WAN_DEVICES="wanpipe1 wanpipe2 wanpipe3 "
>>
>> # cat /etc/wanpipe/wanpipe1.conf
>> #================================================
>> # WANPIPE1 Configuration File
>> #================================================
>> #
>> # Note: This file was generated automatically
>> #       by /usr/local/sbin/setup-sangoma program.
>> #
>> #       If you want to edit this file, it is
>> #       recommended that you use wancfg program
>> #       to do so.
>> #================================================
>> # Sangoma Technologies Inc.
>> #================================================
>>
>>
>>
>>
>> [devices]
>> wanpipe1 = WAN_AFT_ISDN_BRI, Comment
>>
>> [interfaces]
>> w1g1 = wanpipe1, , TDM_VOICE_API, Comment
>>
>> [wanpipe1]
>> CARD_TYPE     = AFT
>> S514CPU     = A
>> CommPort     = PRI
>> AUTO_PCISLOT     = NO
>> PCISLOT     = 0
>> PCIBUS      = 5
>> FE_MEDIA    = BRI
>> FE_LINE        = 1
>> TDMV_LAW    = ALAW
>> RM_BRI_CLOCK_MASTER    = NO
>> MTU         = 1500
>> UDPPORT     = 9000
>> TTL        = 255
>> IGNORE_FRONT_END = NO
>> TDMV_SPAN    = 1
>> TE_AIS_MAINTENANCE = NO            #NO: defualt  YES: Start port in AIS
>> Blue Alarm and keep line down
>>                                    #wanpipemon -i w1g1 -c Ttx_ais_off to
>> disable AIS maintenance mode
>>                                    #wanpipemon -i w1g1 -c Ttx_ais_on to
>> enable AIS maintenance mode
>> TDMV_HW_DTMF        = YES        # YES: receive dtmf events from hardware
>> TDMV_HW_FAX_DETECT      = YES     # YES: receive fax 1100hz events from
>> hardware
>> HWEC_OPERATION_MODE     = OCT_NORMAL    # OCT_NORMAL: echo cancelation
>> enabled with nlp (default)
>>                                            # OCT_SPEECH: improves software
>> tone detection by disabling NLP (echo possible)
>>                                            # OCT_NO_ECHO:disables echo
>> cancelation but allows VQE/tone functions.
>> HWEC_DTMF_REMOVAL       = NO    # NO: default  YES: remove dtmf out of
>> incoming media (must have hwdtmf enabled)
>> HWEC_NOISE_REDUCTION    = NO    # NO: default  YES: reduces noise on the
>> line - could break fax
>> HWEC_ACUSTIC_ECHO       = NO    # NO: default  YES: enables acustic echo
>> cancelation
>> HWEC_NLP_DISABLE        = NO    # NO: default  YES: guarantees software
>> tone detection (possible echo)
>> HWEC_TX_AUTO_GAIN       = 0     # 0: disable   -40-0: default tx audio
>> level to be maintained (-20 default)
>> HWEC_RX_AUTO_GAIN       = 0     # 0: disable   -40-0: default tx audio
>> level to be maintained (-20 default)
>> HWEC_TX_GAIN            = 0     # 0: disable   -24-24: db values to be
>> applied to tx signal
>> HWEC_RX_GAIN            = 0     # 0: disable   -24-24: db values to be
>> applied to tx signal
>>
>> [w1g1]
>> ACTIVE_CH    = ALL
>> TDMV_HWEC    = YES
>> MTU         = 80
>>
>>     ... and wanpipe2 was also automatically generated so just has '2' for
>> FE_LINE and TDMV_SPAN instead of '1'.
>>
>> # cat /usr/local/freeswitch/conf/freetdm.conf
>> [span wanpipe wp1]
>> trunk_type =>   bri
>> group=1
>> b-channel =>   1:1-2
>> d-channel =>   1:3
>>
>> [span wanpipe wp2]
>> trunk_type =>   bri
>> group=1
>> b-channel =>   2:1-2
>> d-channel =>   2:3
>>
>> [span wanpipe FXS]
>> name =>   freetdm
>> trunk_type =>   fxs
>> group =>   grp2
>> fxs-channel =>   3:1
>>
>> trunk_type =>   fxs
>> group =>   grp2
>> fxs-channel =>   3:2
>>
>> # cat /usr/local/freeswitch/conf/autoload_configs/freetdm.conf.xml
>> <configuration name="freetdm.conf" description="Freetdm Configuration">
>> <settings>
>> <param name="debug" value="0"/>
>> <!--<param name="hold-music" value="$${moh_uri}"/>-->
>> <!--<param name="enable-analog-option" value="call-swap"/>-->
>> <!--<param name="enable-analog-option" value="3-way"/>-->
>> </settings>
>>
>> <config_profiles>
>>
>> <profile name="my_bri_te_1">
>> <param name="switchtype" value="euroisdn" />
>> <param name="interface" value="cpe"/>
>> <param name="facility" value="no" />
>> </profile>
>>
>> </config_profiles>
>>
>> <sangoma_bri_spans>
>>
>> <span name="wp1" cfgprofile="my_bri_te_1">
>> <param name="dialplan" value="XML"/>
>> <param name="context" value="public"/>
>> </span>
>>
>> <span name="wp2" cfgprofile="my_bri_te_1">
>> <param name="dialplan" value="XML"/>
>> <param name="context" value="public"/>
>> </span>
>>
>> </sangoma_bri_spans>
>>
>>
>> <analog_spans>
>>
>> <span name="FXS">
>> <!--<param name="hold-music" value="$${moh_uri}"/>-->
>> <param name="dialplan" value="XML"/>
>> <param name="context" value="default"/>
>> <!-- regex to stop dialing when it matches -->
>> <!--<param name="dial-regex" value="5555"/>-->
>> <!-- regex to stop dialing when it does not match -->
>> <!--<param name="fail-dial-regex" value="^5"/>-->
>> </span>
>>
>> </analog_spans>
>>
>> </configuration>
>>
>> What other config files do I need to supply? I don't want to deluge the
>> list any more in my first post!
>>
>> 2. We have 2 Single Number DDI numbers configured over the 4 channels.
>> I want to set up outgoing calls so that they can appear to come from
>> either of the two numbers, but at present the outgoing CLI appears to be
>> overridden by the telco (BT) to only use one of the numbers.  Has anyone
>> got this working in UK, and what is the format for the
>> outbound_caller_id_number: last 6 digits or full 11 digits?  I note that
>> the inbound called number is only the last 6 digits.
>>
>> 3. I have built FreeSWITCH from git and installed at /usr/local/... and
>> then followed the steps on the Ubuntu page in the Wiki to set up the run
>> control scripts, etc, and run FS non-root as freeswitch:daemon.  With
>> FreeTDM, I have discovered that the /dev/wan* devices are owned by
>> root:root, and so are inaccessible to FS running as non-root.  So for
>> now I have added a line in /etc/init.d/freeswitch to 'chgrp freeswitch
>> /dev/wan*'.  This is not the most elegant solution, because 'wanrouter
>> restart' (which seems to be my best friend at the moment) resets the
>> ownership to root:root.  I have tried grepping to see where the mknods
>> are for these devices, but have been unsuccessful.  Is there a better
>> place to 'permanently' change the device ownership?
>>
>> Thanks for all the great support I have already got just from editing my
>> Wiki User page; this is a friendly group!
>>
>> John
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From benkokakao at gmail.com  Fri Jun 10 12:29:47 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Fri, 10 Jun 2011 10:29:47 +0200
Subject: [Freeswitch-users] fetch password
In-Reply-To: <BANLkTimZv5uM8+atzR15qds8Ap3UHVbU6Q@mail.gmail.com>
References: <BANLkTinbM8ZW7DCyBYWp3NoMU8fU72-wQQ@mail.gmail.com>
	<BANLkTimikq6z2T+thzDm8ZDDZx7yFAhUNA@mail.gmail.com>
	<BANLkTimZv5uM8+atzR15qds8Ap3UHVbU6Q@mail.gmail.com>
Message-ID: <BANLkTikEvezO5RH2UuEQy8cdUvZhQFouRw@mail.gmail.com>

On 10 June 2011 09:38, Sam <u2nsam at gmail.com> wrote:
> How Does it access database , does we need to write some syntax somewhere,
> so that FS fetches it out from DB.

Hi!

I'm still not sure if i understand your question.

The default configuration uses a sqlite-DB which is saved in
/usr/local/freeswitch/db/voicemail_default.db. The
voicemail-application accesses this db without additional
configuration.

If you however want to save the voicemail-preferences in a
non-embedded DB like PostgreSQL or MySQL, you have to configure odbc
on your host to access them plus activate odbc in voicemail.conf.xml.

Please read the following pages if that is what you want to do:
http://wiki.freeswitch.org/wiki/Mod_voicemail#odbc-dsn
http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core

hth,
Christian


From benkokakao at gmail.com  Fri Jun 10 12:44:28 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Fri, 10 Jun 2011 10:44:28 +0200
Subject: [Freeswitch-users] Voicemail Message - How t know how
 manyareread & how many are unread in Inbox
In-Reply-To: <DFC9276BB70D4FB99BD4FF88483B2590@sharad>
References: <mailman.10243.1307524956.24555.freeswitch-users@lists.freeswitch.org>
	<AA1BE0FA88F3421784DBD1D91A84E515@sharad>
	<BANLkTikshbtzM67daHBePp7TB5pH5FjEyA@mail.gmail.com>
	<E826C94A3C6BE44198FAF4D66B5AB4D81BDA4D5BF1@VA3DIAXVS351.RED001.local>
	<BANLkTikBYBK5Jugw6dtqUL88bkkic-bFAA@mail.gmail.com>
	<26DE8A8116AB4F5BB31F0675DDBFD1BF@sharad>
	<BANLkTiknapraGXceJXfcf9RHJXBRpbE0QQ@mail.gmail.com>
	<DFC9276BB70D4FB99BD4FF88483B2590@sharad>
Message-ID: <BANLkTi=jMGpHPJNdRLXVvMEP09GuBX5dJg@mail.gmail.com>

> Plz advice further..

Sorry, no idea then.
But i assume, if a user just listens to a message but doesn't archive
it right away, he want's to listen to it again at a later time, so the
message should remain "new" - therefore the behaviour you report is
not wrong imho(not updating "read_epoch" is not beautiful though).


From mariusz_kolo at wp.pl  Fri Jun 10 13:02:20 2011
From: mariusz_kolo at wp.pl (=?UTF-8?B?TWFyaXVzeiBLb8WCb2R6aWVqY3p5aw==?=)
Date: Fri, 10 Jun 2011 11:02:20 +0200
Subject: [Freeswitch-users] Logical OR in conditions
In-Reply-To: <43439B44-F6EE-448F-9CB4-58D15ACAE9F6@mralston.com>
References: <43439B44-F6EE-448F-9CB4-58D15ACAE9F6@mralston.com>
Message-ID: <4DF1DD9C.4080907@wp.pl>

Hi

maybe only one condition something like

^[\*7]21[\*7]([0-9]+)[#9]$ - this allows to use combination of *#79

or

^\*21\*([0-9]+)#$|^7217([0-9]+)9$  - this is more restrict, because u
can use *# or 79 but not 7# or *9

this is solution for OR in your code.



W dniu 2011-06-09 00:44, Matthew Ralston pisze:
> Hi,
>
> What is the correct condition->break value to use in order to create a logical OR with conditions?
>
> I have two extensions which I'd like to merge into one. I'd like the actions to be run if either of the conditions matches:
>
> 		<extension name="call_forward_enable">
> 			<condition field="destination_number" expression="^\*21\*([0-9]+)#$">
> 				<action application="javascript" data="callForwardEnable.js"/>
> 				<action application="hangup"/>
> 			</condition>
> 		</extension>
>
> 		<!-- For phones that won't dial *# numbers, *=7, #=9 -->
> 		<extension name="call_forward_enable">
> 			<condition field="destination_number" expression="^7217([0-9]+)9$">
> 				<action application="javascript" data="callForwardEnable.js"/>
> 				<action application="hangup"/>
> 			</condition>
> 		</extension>
>
> So the result would be something like:
>
> 		<extension name="call_forward_enable">
> 			<condition field="destination_number" expression="^\*21\*([0-9]+)#$" break="on-true" />
> 			<condition field="destination_number" expression="^7217([0-9]+)9$">
> 				<action application="javascript" data="callForwardEnable.js"/>
> 				<action application="hangup"/>
> 			</condition>
> 		</extension>
>
> I can't seem to get it to work and my brain is melting trying to wrap my head around the different break values.
>
> Cheers,
>
> Matt
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From shouldbeq931 at gmail.com  Fri Jun 10 13:08:25 2011
From: shouldbeq931 at gmail.com (shouldbe q931)
Date: Fri, 10 Jun 2011 10:08:25 +0100
Subject: [Freeswitch-users] UK ISDN outgoing CLI (was Re: FreeTDM -
 Sangoma B700 - ISDN connection questions - UK)
In-Reply-To: <4DF10C83.6020803@earthspike.net>
References: <4DED4854.6010203@earthspike.net>
	<BANLkTimP23FVs8Vcg+pbxnfBucrQoJur2w@mail.gmail.com>
	<4DF10C83.6020803@earthspike.net>
Message-ID: <BANLkTikCEKqPTFZLDxF+h940RnmizHYLBA@mail.gmail.com>

On Thu, Jun 9, 2011 at 7:10 PM, John <freeswitch at earthspike.net> wrote:
> [I've forked this thread as it really should have been 3 separate
> threads to start with.]
>
> Thanks,
>
> I have tried every combination I can think of, except 10 digits which I
> will give a go. ?That makes sense as in the general case it would permit
> non-geographical numbers to be presented.
>
> The lines are provided as 2 x ISDN2e with a lead number covering all 4
> channels. ?The NTE8s are labelled 1-2 and 3-4, so not separate
> individual lines, I thnk.
>
> We currently have a Single Number DDI in addition, and hope to increase
> that to 2 very soon; the 3 numbers between them will serve 3 separate
> business units, so we want to be able to present the correct outbound
> CLI from each business unit. ?We also have a remote call forward from
> another number to the lead number, but presenting that really would be
> CLI spoofing, so I am not expecting that to be permissible. But I would
> expect those that are directly allocated to the 2 lines/4 channels to be
> allowed to be presented.
>
> John
>
> On 09/06/11 08:27, shouldbe q931 wrote:
>> On Mon, Jun 6, 2011 at 10:36 PM, John<freeswitch at earthspike.net> ?wrote:
>>> Hello,
>>>
>> <snipped for brevity>
>>> 2. We have 2 Single Number DDI numbers configured over the 4 channels.
>>> I want to set up outgoing calls so that they can appear to come from
>>> either of the two numbers, but at present the outgoing CLI appears to be
>>> overridden by the telco (BT) to only use one of the numbers. ?Has anyone
>>> got this working in UK, and what is the format for the
>>> outbound_caller_id_number: last 6 digits or full 11 digits? ?I note that
>>> the inbound called number is only the last 6 digits.
>>>
>> <snipped for brevity>
>>> Thanks for all the great support I have already got just from editing my
>>> Wiki User page; this is a friendly group!
>>>
>>> John
>>>
>> I can only comment on the BT part.
>>
>> If they are two individual lines, you're out of luck with BT, as they
>> do not allow CLI spoofing, which this would be.
>>
>> I would suggest sending BT the same number of digits that they send
>> you, if that doesn't work try 10 digits. If ?you are not sending the
>> "correct" number of digits for a valid DDI on the trunk, then they
>> will send the lead number for the trunk.
>>
>> Cheers
>>
>> _______________________________________________

I know that this might sound like heresy :-)  but have you checked
spoken to BT and asked them if CLIP and CLOP on the lines ?

I think CLOP is outbound and CLIP is inbound, but I might have it the
wrong way round...


From potxoka at gmail.com  Fri Jun 10 13:31:02 2011
From: potxoka at gmail.com (Antonio)
Date: Fri, 10 Jun 2011 11:31:02 +0200
Subject: [Freeswitch-users] Proxy traffic security
In-Reply-To: <BANLkTincK9zEprZjuMRMo5c6vejuX8_22w@mail.gmail.com>
References: <4DEFF3FA.5050500@gmail.com>
	<BANLkTincK9zEprZjuMRMo5c6vejuX8_22w@mail.gmail.com>
Message-ID: <4DF1E456.1030607@gmail.com>

Hi,

If I had thought also use iptables, but it was to know if exists 
something better to put the ip's (to prevent IP spoofing). Thanks.

Greetings
Anto

El 09/06/11 16:01, Christian Benke escribi?:
> ACLs definitely would not hurt. Also think about setting up a simple
> firewall(e.g. Shorewall or just plain IPTables) to limit the access to
> the ip's which MUST have access(e.g. SIP-Provider IPs), you can never
> be sure if a software(like FreeSWITCH) doesn't have a bug that could
> be exploited.
>
> Cheers,
> Christian


From steveayre at gmail.com  Fri Jun 10 13:46:00 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 10 Jun 2011 10:46:00 +0100
Subject: [Freeswitch-users] Proxy traffic security
In-Reply-To: <4DF1E456.1030607@gmail.com>
References: <4DEFF3FA.5050500@gmail.com>
	<BANLkTincK9zEprZjuMRMo5c6vejuX8_22w@mail.gmail.com>
	<4DF1E456.1030607@gmail.com>
Message-ID: <BANLkTi=MGq5LSn7j+UmDFQLR2p0RHSoPGQ@mail.gmail.com>

Iptables will allow you to block all but your proxies from sending. It
should work (slightly) faster than the FS ACL because it'll block it earlier
in the packet processing.

-Steve


On 10 June 2011 10:31, Antonio <potxoka at gmail.com> wrote:

> Hi,
>
> If I had thought also use iptables, but it was to know if exists
> something better to put the ip's (to prevent IP spoofing). Thanks.
>
> Greetings
> Anto
>
> El 09/06/11 16:01, Christian Benke escribi?:
> > ACLs definitely would not hurt. Also think about setting up a simple
> > firewall(e.g. Shorewall or just plain IPTables) to limit the access to
> > the ip's which MUST have access(e.g. SIP-Provider IPs), you can never
> > be sure if a software(like FreeSWITCH) doesn't have a bug that could
> > be exploited.
> >
> > Cheers,
> > Christian
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From potxoka at gmail.com  Fri Jun 10 13:37:49 2011
From: potxoka at gmail.com (Antonio)
Date: Fri, 10 Jun 2011 11:37:49 +0200
Subject: [Freeswitch-users] Proxy traffic security
In-Reply-To: <BANLkTimsyiaWov8GKL8u+dxmqKPOhHFckw@mail.gmail.com>
References: <4DEFF3FA.5050500@gmail.com>
	<BANLkTimsyiaWov8GKL8u+dxmqKPOhHFckw@mail.gmail.com>
Message-ID: <4DF1E5ED.2080509@gmail.com>

Hi,

Thank you very much for the reply, I will look at this solution.

Best regards
Anto

El 09/06/11 16:17, Steven Ayre escribi?:
> You can create an ACL listing proxy IPs then set the proxy-acl 
> parameter on the SIP profile.call
>
> If your proxy can add a X-AUTH-IP header to the INVITE containing the 
> caller's IP then FS will use that IP to check against ACLs instead of 
> the IP received from, if the IP received from is on the proxy ACL. 
> That way you can still authenticate callers with ACLs behind a proxy, 
> although you're trusting your proxy to set that header correctly. The 
> proxy-acl setting means only your proxy can set the X-AUTH-IP header, 
> it'll be ignored on calls from any other IP.
>
> -Steve



From freeswitch at earthspike.net  Fri Jun 10 14:22:50 2011
From: freeswitch at earthspike.net (John)
Date: Fri, 10 Jun 2011 11:22:50 +0100
Subject: [Freeswitch-users] UK ISDN outgoing CLI (was Re: FreeTDM -
 Sangoma B700 - ISDN connection questions - UK)
In-Reply-To: <BANLkTikCEKqPTFZLDxF+h940RnmizHYLBA@mail.gmail.com>
References: <4DED4854.6010203@earthspike.net>	<BANLkTimP23FVs8Vcg+pbxnfBucrQoJur2w@mail.gmail.com>	<4DF10C83.6020803@earthspike.net>
	<BANLkTikCEKqPTFZLDxF+h940RnmizHYLBA@mail.gmail.com>
Message-ID: <4DF1F07A.5000200@earthspike.net>

Shouldbe,

You won't be burned at the stake for heresy and are probably right that 
this is the problem, as we have had delays in getting CLIP implemented. 
 From your comment I assume that COLP is not automatically enabled (even 
though I am paying for additional SNDDIs) but has to be ordered 
specifically.  Our service provider is on to it, and I am waiting till 
CLIP/COLP is applied before doing any more testing.  I'll post here with 
the results.

John

On 10/06/11 10:08, shouldbe q931 wrote:
> On Thu, Jun 9, 2011 at 7:10 PM, John<freeswitch at earthspike.net>  wrote:
>> [I've forked this thread as it really should have been 3 separate
>> threads to start with.]
>>
>> Thanks,
>>
>> I have tried every combination I can think of, except 10 digits which I
>> will give a go.  That makes sense as in the general case it would permit
>> non-geographical numbers to be presented.
>>
>> The lines are provided as 2 x ISDN2e with a lead number covering all 4
>> channels.  The NTE8s are labelled 1-2 and 3-4, so not separate
>> individual lines, I thnk.
>>
>> We currently have a Single Number DDI in addition, and hope to increase
>> that to 2 very soon; the 3 numbers between them will serve 3 separate
>> business units, so we want to be able to present the correct outbound
>> CLI from each business unit.  We also have a remote call forward from
>> another number to the lead number, but presenting that really would be
>> CLI spoofing, so I am not expecting that to be permissible. But I would
>> expect those that are directly allocated to the 2 lines/4 channels to be
>> allowed to be presented.
>>
>> John
>>
>> On 09/06/11 08:27, shouldbe q931 wrote:
>>> On Mon, Jun 6, 2011 at 10:36 PM, John<freeswitch at earthspike.net>    wrote:
>>>> Hello,
>>>>
>>> <snipped for brevity>
>>>> 2. We have 2 Single Number DDI numbers configured over the 4 channels.
>>>> I want to set up outgoing calls so that they can appear to come from
>>>> either of the two numbers, but at present the outgoing CLI appears to be
>>>> overridden by the telco (BT) to only use one of the numbers.  Has anyone
>>>> got this working in UK, and what is the format for the
>>>> outbound_caller_id_number: last 6 digits or full 11 digits?  I note that
>>>> the inbound called number is only the last 6 digits.
>>>>
>>> <snipped for brevity>
>>>> Thanks for all the great support I have already got just from editing my
>>>> Wiki User page; this is a friendly group!
>>>>
>>>> John
>>>>
>>> I can only comment on the BT part.
>>>
>>> If they are two individual lines, you're out of luck with BT, as they
>>> do not allow CLI spoofing, which this would be.
>>>
>>> I would suggest sending BT the same number of digits that they send
>>> you, if that doesn't work try 10 digits. If  you are not sending the
>>> "correct" number of digits for a valid DDI on the trunk, then they
>>> will send the lead number for the trunk.
>>>
>>> Cheers
>>>
>>> _______________________________________________
> I know that this might sound like heresy :-)  but have you checked
> spoken to BT and asked them if CLIP and CLOP on the lines ?
>
> I think CLOP is outbound and CLIP is inbound, but I might have it the
> wrong way round...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From peter.olsson at visionutveckling.se  Fri Jun 10 14:44:40 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 10 Jun 2011 12:44:40 +0200
Subject: [Freeswitch-users] how to redirect $${sounds_dir} ?
In-Reply-To: <BANLkTinv+-0wP0NPyAVTRsG595HPCmUObtTNfy8m0o7f7_Gspw@mail.gmail.com>
References: <BANLkTinv+-0wP0NPyAVTRsG595HPCmUObtTNfy8m0o7f7_Gspw@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59DEE587AA@cooper>

This is defined in vars.xml in the conf folder.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r jesse
Skickat: den 9 juni 2011 23:31
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] how to redirect $${sounds_dir} ?

I would like to pint sounds_dir to a different location, but it seems freeswitch doesn't support --sounds_dir  as command line arguments.
please let me know how to re-define sounds_dir?

thanks!

!DSPAM:4df13db632761312720727!
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From kris at kriskinc.com  Fri Jun 10 17:40:31 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Fri, 10 Jun 2011 09:40:31 -0400
Subject: [Freeswitch-users] CISCO 2811 - FreeSwitch - IVR
In-Reply-To: <A0AF161BE92F1B4CB28FDCBF2524B0420B782B@prdexch01.wnr.internal>
References: <A0AF161BE92F1B4CB28FDCBF2524B0420B782B@prdexch01.wnr.internal>
Message-ID: <BANLkTi=N7SZv5uj5ZqSNE8vr8vyDqm_FJw@mail.gmail.com>

Oliver,

  This is relatively common with POTS interfaces.  Here's an old Cisco
article but it should still be relevant:

http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00800ae2d1.shtml

On Fri, Jun 10, 2011 at 12:19 AM, Schenk, Oliver <OSchenk at wnr.com.au> wrote:
> Hey all,
>
>
>
> I have successfully managed the following setup ? at least to a certain
> extent.
>
>
>
> Phone line -> CISCO VIC2-4FXO -> CISCO 2811 -> Sip -> Windows Server 2008 ->
> FreeSwitch -> Managed IVR (IAppPlugin)
>
>
>
> When I dial say the phone line?s number (49794) it connects me to extension
> 1024 which has the managed C#.NET IVR sitting behind it. That works fine
> while everything is happy. However, when I hang up my phone. My IVR is still
> happily talking to no one until the relevant timeouts expire. Is this the
> normal way to implement it? Just wait for timeout? Or do I have some CISCO
> configuration issues?
>
>
>
> Shouldn?t the CISCO tell FreeSwitch that ?Hey ? the guy on the line has put
> down his phone receiver so you can end your session now.??
>
>
>
>
>
> Any comments?
>
>
>
> voice-port 0/3/0
>
> ?input gain 10
>
> ?output attenuation 10
>
> ?no comfort-noise
>
> ?cptone AU
>
> ?connection plar opx 1024
>
> ?impedance complex1
>
> !
>
> voice-port 0/3/1
>
> !
>
> voice-port 0/3/2
>
> !
>
> voice-port 0/3/3
>
> !
>
> !
>
> !
>
> !
>
> !
>
> dial-peer voice 2 voip
>
> ?destination-pattern .T
>
> ?session protocol sipv2
>
> ?session target sip-server
>
> ?session transport udp
>
> ?dtmf-relay rtp-nte
>
> ?codec g711ulaw
>
> ?no vad
>
> !
>
> !
>
> sip-ua
>
> ?credentials username 1000 password * ******* realm default
>
> ?authentication username 1000 password * ********
>
> ?retry invite 2
>
> ?retry register 10
>
> ?timers connect 100
>
> ?registrar ipv4:192.168.255.**:5060 expires 3600
>
> ?sip-server ipv4:192.168.255.**:5060
>
> !
>
>
>
> Regards,
>
>
>
>
>
> Oliver Schenk
>
>
>
> Control Systems Engineer
>
>
>
> WestNet Rail
>
>
>
> NOTICE - This e-mail and any files transmitted with it are confidential and
> are only for the use of the person to whom they are addressed.
> If you are not the intended recipient then you have received this e-mail in
> error; please advise us immediately if this is the case.
> Any views expressed in this message are those of the individual sender,
> except where the sender specifically states them to be the views of WestNet
> Rail.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner


From Hector.Geraldino at ip-soft.net  Fri Jun 10 18:00:02 2011
From: Hector.Geraldino at ip-soft.net (Hector Geraldino)
Date: Fri, 10 Jun 2011 10:00:02 -0400
Subject: [Freeswitch-users] originating a call from an outbound
	socket	app
In-Reply-To: <BANLkTikG8TQ+1JhAVr8Y6xMWhH1xqor8zA@mail.gmail.com>
References: <6A6B4C284AD15042B429EB9D904544AD021F45EAA0@NY1-EXMB-01.ip-soft.net>
	<BANLkTi=LX=cGqSVrT6o-0OOPXPOA2U=MnA@mail.gmail.com>
	<6A6B4C284AD15042B429EB9D904544AD021F45EB71@NY1-EXMB-01.ip-soft.net>
	<BANLkTikG8TQ+1JhAVr8Y6xMWhH1xqor8zA@mail.gmail.com>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD021F45EBBC@NY1-EXMB-01.ip-soft.net>

Thanks for your response Michael.

Finally I found the way of dialing to an external extension. What I did was just open a shell, listen in a port using nc  and tried to originate a call. It worked using the syntax:

originate sofia/internal/5562 at 192.168.23.104 &park

As this is an outbound call (not managed by the socket application) I had to use a new inbound socket connection to connect to FS, make the call, query for the uuid and receive some events (like DTMF among others).  Everything was going great, until I found two issues:


-          The CHANNEL_ANSWER event is always triggered, no matter if the call is actually picked up or not on my phone. I thought that maybe I can use some other event to determine if the call have been answered or not (like CHANNEL_CALLSTATE or CALL_UPDATE), but the additional information on the event headers is the same no matter if it's answered or not.

-          When the call is answered, and the user presses a key to accept the call, the function returns the UUID of this session to the main method. Then I tried to make the bridge between the two calls (the legA call handled by the socket application, and the legB outbound call generated in the application using an inbound connection). I using the command

api uuid_bridge <incoming_uuid> <outbound_uuid>

but this didn't work. Am I missing something here?

Thanks again,
Hector


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Thursday, June 09, 2011 6:13 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] originating a call from an outbound socket app

Also, I don't think this is helping you:
  EslMessage response = sendSyncSingleLineCommand(channel, "myevents");

Look here to see what "myevents" does:
http://wiki.freeswitch.org/wiki/Mod_event_socket#Special_Case_-_.27myevents.27
-MC

On Thu, Jun 9, 2011 at 1:22 PM, Hector Geraldino <Hector.Geraldino at ip-soft.net<mailto:Hector.Geraldino at ip-soft.net>> wrote:
Hi Michael,

Thanks for replying back. I've pasted a sample application on pastebin, the URL is:

http://pastebin.com/MYusWe3V

What I'm trying to do (and have partially accomplished) is:

-Answer the call
- Play a message
- wait for a command (dtmf or voice, this is out of the sample app for the sake of keeping it short & simple)
- originate a new call, wait for it to be answered and depending on an event (dtmf)...
- ...do the bridge or dial another extension.

The extension number is external (managed by a Cisco CCM). If I try to do the bridge from the XML dialplan it works perfect. When I try to do it from an outbound socket java application, anything happens, and almost no information is logged on FS log that I can use to decide what might be happening.

Thanks again for your help. I truly appreciate it.

Hector

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
Sent: Thursday, June 09, 2011 12:47 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] originating a call from an outbound socket app

I would say that you want to bridge the inbound leg to the outbound leg and use the answer confirmation variables. I recommend you pastebin a simple java app that answers the call and attempts to bridge it and also pastebin the console log. The community will take a look and see if we can help you figure out what is happening.

-MC
On Wed, Jun 8, 2011 at 3:14 PM, Hector Geraldino <Hector.Geraldino at ip-soft.net<mailto:Hector.Geraldino at ip-soft.net>> wrote:
Hi all,

I've been playing with FS around weeks, with some level of success. Right now I'm stuck trying to dial an extension, or to bridge a call to an external extension. Briefly, the scenario is the following:


-          Cisco CCM forwarding SIP calls to FS

-          FS in outbound mode,  forwarding calls to a Java application using freeswitch-esl jar

The java application answers the call, says a dialogue using TTS and receives input using DTFM or speech commands; all this is working pretty well. Now I'm trying to put this call in hold, originate a call to another extension and, depending on the result (call answered, not answered, destination user presses an specific number generating an expected DTMF), I need to bridge both calls and finish the session.

I try several ways to get this done with no luck yet: using api originate to make the call with different arguments ({ignore_early_media,  etc). I even tried to skip this step and just do the bridge between the inbound call and the external extension, but this  didn't work either. I can't see any errors thrown by FS, and the result of the command is +OK, but nothing happens, the external phone never rings. The FS log only says:

[DEBUG] switch_core_session.c:954 Send signal sofia/external/5655 at 192.168.8.1<mailto:5655 at 192.168.8.1> [BREAK] <= this ext/IP address is from the inbound caller, not the one I'm trying to connect to.

My first impression was that I might have something wrong with the originate/bridge parameters. I changed the dialplan XML file to just do the bridge, and it worked like a charm.

<include>
<extension name="test">
        <condition field="destination_number" expression="6114|5121">
<!--    <action application="socket" data="localhost:8044 async full"/>  -->
          <action application="bridge" data="sofia/external/5562 at 192.168.2.14<mailto:5562 at 192.168.2.14>" />
        </condition>
</extension>
</include>

I don't know what path to follow here. I was unable to find relevant examples on google that guides me to solve this little issue. Any advices?

Thanks all,
Hector

_______________________________________________
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_______________________________________________
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From freeswitch-list at puzzled.xs4all.nl  Fri Jun 10 20:38:44 2011
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Fri, 10 Jun 2011 18:38:44 +0200
Subject: [Freeswitch-users] FreeTDM - Sangoma B700 - ISDN connection
 questions - UK
In-Reply-To: <4DF1CF66.4050700@earthspike.net>
References: <4DED4854.6010203@earthspike.net> <4DF1CAB8.3070706@gmail.com>
	<4DF1CF66.4050700@earthspike.net>
Message-ID: <4DF24894.4090609@puzzled.xs4all.nl>

On 06/10/2011 10:01 AM, John wrote:
[snip]
> I've also resolved the device permissions issue with help from Sangoma
> support, and will post on that/update the Ubuntu Wiki page when I get a
> moment, probably after the weekend.

Can you please post here also what those permission issues were and how 
you solved them so people like me who don't use Ubuntu and have no idea 
which Ubuntu Wiki you refer too can learn from this too?

Or possibly add it to the appropriate section on the FreeSWITCH Wiki at
http://wiki.freeswitch.org

Thanks and regards,
Patrick


From chat2jesse at gmail.com  Fri Jun 10 21:40:18 2011
From: chat2jesse at gmail.com (jesse)
Date: Fri, 10 Jun 2011 10:40:18 -0700
Subject: [Freeswitch-users] how to redirect $${sounds_dir} ?
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59DEE587AA@cooper>
References: <BANLkTinv+-0wP0NPyAVTRsG595HPCmUObtTNfy8m0o7f7_Gspw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DEE587AA@cooper>
Message-ID: <BANLkTikiLd0OWyHMz0znc14MNH79HAOzSg@mail.gmail.com>

vars.xml only sets sound_prefix :

 <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/>

$${sounds_dir} is like $${local_ip_v4}, automatically fetched by the
application.

not sure whether i can set sounds_dir in vars.xml...



On Fri, Jun 10, 2011 at 3:44 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> This is defined in vars.xml in the conf folder.
>
>
>
> /Peter
>
>
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *jesse
> *Skickat:* den 9 juni 2011 23:31
> *Till:* FreeSWITCH Users Help
> *?mne:* [Freeswitch-users] how to redirect $${sounds_dir} ?
>
>
>
> I would like to pint sounds_dir to a different location, but it seems
> freeswitch doesn't support --sounds_dir  as command line arguments.
>
> please let me know how to re-define sounds_dir?
>
>
>
> thanks!
>
>
>
> !DSPAM:4df13db632761312720727!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From joaocarlosleme at gmail.com  Sat Jun 11 00:26:05 2011
From: joaocarlosleme at gmail.com (Joao Leme)
Date: Fri, 10 Jun 2011 13:26:05 -0700
Subject: [Freeswitch-users] FIFO suddenly stopped working properly
In-Reply-To: <BANLkTik2QK=SH1Yjg1DPBVKbr7HNqLdD8g@mail.gmail.com>
References: <BANLkTim8MJUDxc+0aQusxckV4+X6mrHwDA@mail.gmail.com>
	<BANLkTin72-yCR=gzG1+trJ3i63h_kzsn_A@mail.gmail.com>
	<BANLkTik2QK=SH1Yjg1DPBVKbr7HNqLdD8g@mail.gmail.com>
Message-ID: <BANLkTikVBn88USxFXpWMTv33qGPPr=HKdw@mail.gmail.com>

Anyone?

Anyone else having the same issue?

On Fri, Jun 3, 2011 at 3:31 PM, Joao Leme <joaocarlosleme at gmail.com> wrote:

> Thanks Michael, but I don't know what I have to do to get it fixed. Is
> there a configuration change? I'm not good at debugging and probably I
> should change the subject to "How to configure FIFO properly" (hoping to be
> a configuration problem).
>
> Below is my simple configuration for FIFO:
>
> fifo.conf.xml:
> <fifo name="sales_fifo@$${domain}" importance="0">
>       <member timeout="10" simo="10"
> lag="3">{fifo_member_wait=nowait}group/attendants@$${domain}</member>
>       <member timeout="300" simo="10"
> lag="3">{fifo_member_wait=nowait}group/sales@$${domain}</member>
> </fifo>
>
> directory/default.xml:
> <groups>
>  <group name="sales">
>         <users>
>           <user id="1000" type="pointer"/>
>           <user id="1001" type="pointer"/>
>           <user id="1002" type="pointer"/>
>           <user id="1003" type="pointer"/>
>         </users>
>       </group>
>  <group name="attendants">
>         <users>
>           <user id="1002" type="pointer"/>
>           <user id="1003" type="pointer"/>
>         </users>
>       </group>
> </groups>
>
> dialplan/default/mydialplan.xml: putting call on fifo
> <extension name="sales_fifo_in">
>       <condition field="destination_number" expression="^sales$" />
>   <action application="answer"/>
>     <action application="set" data="fifo_chime_freq=30"/>
> <action application="set" data="fifo_orbit_dialplan=XML"/>
>  <action application="set" data="fifo_orbit_context=default"/>
>  <action application="set" data="fifo_caller_exit_key=0"/>
>  <action application="set" data="fifo_caller_exit_to_orbit=true"/>
>           <action application="set"
> data="fifo_music=$${base_dir}/sounds/music/8000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav"/>
>  <action application="fifo" data="sales_fifo@$${domain} in undef"/>
>     </extension>
>
> It used to work but nows it only sends one call at a time, even with the
> other users available. I also tried replacing the groups for users but no
> luck.
>
> Thanks
>
> On Fri, Jun 3, 2011 at 12:53 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Also, check out this thread, particularly the comments from Anthony:
>>
>> http://lists.freeswitch.org/pipermail/freeswitch-users/2011-March/070588.html
>>
>> If you had a really old version of FIFO and updated then you may have
>> crossed over from when caller ID was not the default to the current state
>> where it is the default.
>>
>> -MC
>>
>> On Thu, Jun 2, 2011 at 2:02 PM, Joao Leme <joaocarlosleme at gmail.com>wrote:
>>
>>> Hi there,
>>> No change nor updates have been made in months. Everything was working
>>> great but now FIFO is not ringing the extensions while the first call is not
>>> HUNG UP...meaning, first call to go to FIFO rings the agents, and one
>>> person answer...when the second call comes in, it goes to fifo queue (caller
>>> listens the music) but no one is aware of the call, nor even the agents that
>>> are available. Only after the first call hangs up the 2nd goes in. I tried
>>> troubleshooting but no help.
>>>
>>> What I also noticed is that before, to display the caller ID on my sip
>>> client I had to use:
>>>
>>> <action application="set"
>>> data="origination_caller_id_number=${caller_id_number}"/>
>>> <action application="set"
>>> data="origination_caller_id_name=${caller_id_name}"/>
>>>
>>> but now, while testing, the caller id was showing when calling
>>> fifo without setting those parameters, but before it would just show the
>>> fifo queue name.
>>>
>>> All I can say is that I can't figure out why and how to fix these
>>> behavior if I have done no change to FreeSWITCH nor the config files. I also
>>> tried downloading the latest GIT and got the same response.
>>>
>>> THANKS,
>>> John
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From lrobot.qq at gmail.com  Fri Jun 10 13:06:35 2011
From: lrobot.qq at gmail.com (lrobot)
Date: Fri, 10 Jun 2011 17:06:35 +0800
Subject: [Freeswitch-users] Require help for 2833 dtmf problem( application:
	bridge, bypass_media=true)
Message-ID: <BANLkTim+4LRBX3ds1tKYUeOoV4gi0BM2zQ@mail.gmail.com>

Hi Freeswitch gay,

  I use freeswitch to bridge one incoming call to other target phone number.
  I use bypass_media mode.
  So the rtp is not go through the freeswitch,
  By I notice that When LegB receive 183 or 200 SDP message. Freeswitch
change it 2833 telephone-event payload type. and forwarding the message to
caller. That make the caller use wrong dtmf payload type to send dtmf

  What wrong with that?  How can fix it?
  Any information or help is welcome.


Thanks


Lrobot
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From stephenl at yourgolftravel.com  Fri Jun 10 21:27:23 2011
From: stephenl at yourgolftravel.com (Stephen Lewis)
Date: Fri, 10 Jun 2011 18:27:23 +0100
Subject: [Freeswitch-users] FreeTDM - Sangoma B700 - ISDN connection
 questions - UK
In-Reply-To: <4DF1CF66.4050700@earthspike.net>
References: <4DED4854.6010203@earthspike.net> <4DF1CAB8.3070706@gmail.com>
	<4DF1CF66.4050700@earthspike.net>
Message-ID: <BANLkTinC2xZwwWxs6cbU+8=GTKsq8uansQ@mail.gmail.com>

Hi John

Would be interesting to hear whether you did something along these
lines - creating /etc/udev/rules.d/80-wanpipe.rules containing

SUBSYSTEM=="wanpipe", OWNER="freeswitch", GROUP="daemon"
SUBSYSTEM=="wp_ec", OWNER="freeswitch", GROUP="daemon"

That's what we're doing on Ubuntu 10.04.2, and it seems to work.

I assume the solution currently described in the wiki[0] doesn't
survive a reboot?

Thanks

Stephen

[0] http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start#Using_FreeTDM_with_the_.27direct_from_git.27_method_above

On 10 June 2011 09:01, John <freeswitch at earthspike.net> wrote:
> Nazim,
>
> Late last night this got resolved; sorry I didn't post then. ?We have
> had problems with the second line, and when these were fixed last night,
> the problem disappeared. ?So now I have a nice stable FreeSWITCH with
> the only outputs on /log 7 being SIP re-registrations. ?So, you are
> probably correct, and it was a line quality issue.
>
> I've also resolved the device permissions issue with help from Sangoma
> support, and will post on that/update the Ubuntu Wiki page when I get a
> moment, probably after the weekend.
>
> Thanks,
>
> John
>
> On 10/06/11 08:41, Nazim Aghabayov wrote:
>> Hello John,
>>
>> I'm not an ISDN expert, but have some experience with PRI.
>> Have you checked the line? Sometimes bad physical interconnects makes
>> line noisy.
>> Channels going up and down may indicate a physical line problem or
>> inadequate grounding (or mixed signal / common grounds). I would disable
>> hardware echo-canellation and hw. dtmf during the troubleshooting.
>> On PRI it's possible to trace the D channel with "wanpipemon -i w1g1
>> -pcap -pcap_file isdn.pcap -prot ISDN -full -systime -c trd". It should
>> be supported on BRI too, look through wanpipemon manual.
>>
>> Best Regards,
>> Nazim
>>
>>
>> On 06/07/2011 02:36 AM, John wrote:
>>> Hello,
>>>
>>> I have just set up a FreeSWITCH box with a Sangoma B700 connected to 2 x
>>> ISDN2e lines (each 2B+D, so 4 voice channels) in UK, so supplied by BT
>>> Openreach. ?There are a number of anomalies that I am trying to solve.
>>> [One of these is that 1 line is dead, but that is for BT Openreach to
>>> resolve rather than anyone on this list.]
>>>
>>> I have a few questions, but as they are all related, I hope you don't
>>> mind them in one post. ?Some basics first. ?The box is an Atom dual-core
>>> with 2GB of memory and a Sangoma B700 card. ?It's built with Ubuntu
>>> 10.04.2 LTS server 64-bit, patched and up to date, and also runs dhcpd,
>>> lighttpd and sshd. ?I built the Sangoma ISDN libraries and FreeSWITCH
>>> using the latest git versions I could ('make current' about 2 weeks
>>> ago). ?We have incoming and outgoing calls working, but some incoming
>>> calls ring in the caller's ear, but nothing appears on the FreeSWITCH
>>> console, and others fail. ?Rebooting the server cures this. ?We have had
>>> problems with lines being disconnected and then reconnected, and it
>>> seems that FreeTDM/wanrouter/whatever doesn't recognise previously
>>> disconnected lines coming back into use, because a reboot finds lines
>>> that were previously reported disconnected ('wanrouter status' or 'ftdm
>>> list'). ?We have ongoing problems with one line that is 'disconnected'
>>> (wanpipe2/wp2) but the queries I am describing below apply equally when
>>> both lines are connected and working.
>>>
>>> 1. There seems to be a lot of ISDN 'chatter' with channels going up and
>>> down all the time even when the switch is completely idle. ?Is this
>>> normal? ?Here is my /log 7 showing two of the cycles (which appear to be
>>> about 50s apart):
>>>
>>> 2011-06-06 22:01:54.992467 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:681
>>> [SNGISDN Q921] wp1: Protocol: Data Link connection DN(4): Disconnect
>>> initiated(263)
>>> 2011-06-06 22:02:29.952464 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:748
>>> [SNGISDN Q931] s1: Interface: Down(261): Dchan(285)
>>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
>>> [s1c1][1:1] Signalling link status changed to DOWN
>>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
>>> [s1c2][1:2] Signalling link status changed to DOWN
>>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
>>> [s1c3][1:3] Signalling link status changed to DOWN
>>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:200
>>> [s1c1][1:1] Setting availability rate to:5
>>> 2011-06-06 22:02:29.952464 [DEBUG] mod_freetdm.c:2268 got clear channel
>>> sig [SIGSTATUS_CHANGED]
>>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:186
>>> [s1c1][1:1] Setting availability rate to:5
>>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:200
>>> [s1c2][1:2] Setting availability rate to:5
>>> 2011-06-06 22:02:29.952464 [INFO] mod_freetdm.c:2345 1:1 signalling
>>> changed to :DOWN
>>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:186
>>> [s1c2][1:2] Setting availability rate to:5
>>> 2011-06-06 22:02:29.952464 [DEBUG] mod_freetdm.c:2268 got clear channel
>>> sig [SIGSTATUS_CHANGED]
>>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:200
>>> [s1c3][1:3] Setting availability rate to:5
>>> 2011-06-06 22:02:29.952464 [INFO] mod_freetdm.c:2345 1:2 signalling
>>> changed to :DOWN
>>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:186
>>> [s1c3][1:3] Setting availability rate to:5
>>> 2011-06-06 22:02:29.952464 [DEBUG] mod_freetdm.c:2268 got clear channel
>>> sig [SIGSTATUS_CHANGED]
>>> 2011-06-06 22:02:29.952464 [INFO] mod_freetdm.c:2345 1:3 signalling
>>> changed to :DOWN
>>> 2011-06-06 22:02:29.952464 [INFO] ftmod_sangoma_isdn_stack_rcv.c:631
>>> Received RESTART CFM (dChan:1 ces:0 type:1)
>>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:1179
>>> Receved RESTART, but Restart Indicator IE not present
>>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:681
>>> [SNGISDN Q921] wp1: Protocol: Data Link connection UP(3): UA frame with
>>> F-bit = 1(258)
>>> 2011-06-06 22:02:29.972464 [INFO] ftmod_sangoma_isdn_stack_rcv.c:631
>>> Received RESTART CFM (dChan:1 ces:0 type:0)
>>> 2011-06-06 22:02:29.972464 [INFO] ftmod_sangoma_isdn_stack_rcv.c:729
>>> [SNGISDN Q931] s1: Interface: UP(260): Dchan(285)
>>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
>>> [s1c1][1:1] Signalling link status changed to UP
>>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
>>> [s1c2][1:2] Signalling link status changed to UP
>>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
>>> [s1c3][1:3] Signalling link status changed to UP
>>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:200
>>> [s1c1][1:1] Setting availability rate to:10
>>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:186
>>> [s1c1][1:1] Setting availability rate to:10
>>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:200
>>> [s1c2][1:2] Setting availability rate to:10
>>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:186
>>> [s1c2][1:2] Setting availability rate to:10
>>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:200
>>> [s1c3][1:3] Setting availability rate to:10
>>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:186
>>> [s1c3][1:3] Setting availability rate to:10
>>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:1179
>>> Receved RESTART, but Restart Indicator IE not present
>>> 2011-06-06 22:02:29.972464 [DEBUG] mod_freetdm.c:2268 got clear channel
>>> sig [SIGSTATUS_CHANGED]
>>> 2011-06-06 22:02:29.972464 [INFO] mod_freetdm.c:2345 1:1 signalling
>>> changed to :UP
>>> 2011-06-06 22:02:29.972464 [DEBUG] mod_freetdm.c:2268 got clear channel
>>> sig [SIGSTATUS_CHANGED]
>>> 2011-06-06 22:02:29.972464 [INFO] mod_freetdm.c:2345 1:2 signalling
>>> changed to :UP
>>> 2011-06-06 22:02:29.972464 [DEBUG] mod_freetdm.c:2268 got clear channel
>>> sig [SIGSTATUS_CHANGED]
>>> 2011-06-06 22:02:29.972464 [INFO] mod_freetdm.c:2345 1:3 signalling
>>> changed to :UP
>>> 2011-06-06 22:03:19.912462 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:681
>>> [SNGISDN Q921] wp1: Protocol: Data Link connection DN(4): Disconnect
>>> initiated(263)
>>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:681
>>> [SNGISDN Q921] wp1: Protocol: Data Link connection UP(3): UA frame with
>>> F-bit = 1(258)
>>> 2011-06-06 22:03:19.932483 [INFO] ftmod_sangoma_isdn_stack_rcv.c:631
>>> Received RESTART CFM (dChan:1 ces:0 type:0)
>>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:1179
>>> Receved RESTART, but Restart Indicator IE not present
>>> 2011-06-06 22:03:19.932483 [INFO] ftmod_sangoma_isdn_stack_rcv.c:729
>>> [SNGISDN Q931] s1: Interface: UP(260): Dchan(285)
>>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
>>> [s1c1][1:1] Signalling link status changed to UP
>>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
>>> [s1c2][1:2] Signalling link status changed to UP
>>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
>>> [s1c3][1:3] Signalling link status changed to UP
>>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:200
>>> [s1c1][1:1] Setting availability rate to:10
>>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:186
>>> [s1c1][1:1] Setting availability rate to:10
>>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:200
>>> [s1c2][1:2] Setting availability rate to:10
>>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:186
>>> [s1c2][1:2] Setting availability rate to:10
>>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:200
>>> [s1c3][1:3] Setting availability rate to:10
>>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:186
>>> [s1c3][1:3] Setting availability rate to:10
>>> 2011-06-06 22:03:19.932483 [DEBUG] mod_freetdm.c:2268 got clear channel
>>> sig [SIGSTATUS_CHANGED]
>>> 2011-06-06 22:03:19.932483 [INFO] mod_freetdm.c:2345 1:1 signalling
>>> changed to :UP
>>> 2011-06-06 22:03:19.932483 [DEBUG] mod_freetdm.c:2268 got clear channel
>>> sig [SIGSTATUS_CHANGED]
>>> 2011-06-06 22:03:19.932483 [INFO] mod_freetdm.c:2345 1:2 signalling
>>> changed to :UP
>>> 2011-06-06 22:03:19.932483 [DEBUG] mod_freetdm.c:2268 got clear channel
>>> sig [SIGSTATUS_CHANGED]
>>> 2011-06-06 22:03:19.932483 [INFO] mod_freetdm.c:2345 1:3 signalling
>>> changed to :UP
>>> freeswitch at internal> ? ftdm list
>>> +OK
>>> span: 1 (wp1)
>>> type: Sangoma (ISDN)
>>> physical_status: ok
>>> signaling_status: UP
>>> chan_count: 3
>>> dialplan: XML
>>> context: public
>>> dial_regex:
>>> fail_dial_regex:
>>> hold_music:
>>> analog_options: none
>>> +OK
>>> span: 2 (wp2)
>>> type: Sangoma (ISDN)
>>> physical_status: alarmed
>>> signaling_status: DOWN
>>> chan_count: 3
>>> dialplan: XML
>>> context: public
>>> dial_regex:
>>> fail_dial_regex:
>>> hold_music:
>>> analog_options: none
>>> +OK
>>> span: 3 (FXS)
>>> type: analog
>>> physical_status: ok
>>> signaling_status: UP
>>> chan_count: 2
>>> dialplan: XML
>>> context: default
>>> dial_regex:
>>> fail_dial_regex:
>>> hold_music:
>>> analog_options: none
>>>
>>>
>>> # wanrouter status
>>>
>>> Devices currently active:
>>> ? ? ? ?wanpipe1 wanpipe2 wanpipe3
>>>
>>>
>>> Wanpipe Config:
>>>
>>> Device name | Protocol Map | Adapter ?| IRQ | Slot/IO | If's | CLK |
>>> Baud rate |
>>> wanpipe1 ? ?| N/A ? ? ? ? ?| A500/B700| 20 ?| 0 ? ? ? | 1 ? ?| N/A |
>>> 0 ? ? ? ? |
>>> wanpipe2 ? ?| N/A ? ? ? ? ?| A500/B700| 20 ?| 0 ? ? ? | 1 ? ?| N/A |
>>> 0 ? ? ? ? |
>>> wanpipe3 ? ?| N/A ? ? ? ? ?| A200/A400/B600/B700/B800| 20 ?| 0 ? ? ? |
>>> 1 ? ?| N/A | 0 ? ? ? ? |
>>>
>>> Wanrouter Status:
>>>
>>> Device name | Protocol | Station | Status ? ? ? ?|
>>> wanpipe1 ? ?| AFT ISDN | N/A ? ? | Connected ? ? |
>>> wanpipe2 ? ?| AFT ISDN | N/A ? ? | Disconnected ?|
>>> wanpipe3 ? ?| A-ANALOG | N/A ? ? | Connected ? ? |
>>>
>>>
>>> # cat /etc/wanpipe/wanrouter.rc
>>> #!/bin/sh
>>> # .. comments snipped ...
>>> ROUTER_BOOT=YES
>>> WAN_CONF_DIR=/etc/wanpipe
>>> WAN_INTR_DIR=/etc/wanpipe/interfaces
>>> WAN_BIN_DIR=/usr/sbin
>>> WAN_LOG=/var/log/wanrouter
>>> WAN_LOCK=/var/lock/wanrouter
>>> WAN_LOCK_DIR=/var/lock
>>> WAN_IP_FORWARD=NO
>>> NEW_IF_TYPE=NO
>>> WAN_LIB_DIR=/etc/wanpipe/lib
>>> WAN_ADSL_LIST=/etc/wanpipe/wan_adsl.list
>>> WAN_ANNEXG_LOAD=NO
>>> WAN_SCTP_LOAD=NO
>>> WAN_LIP_LOAD=NO
>>> WAN_DYN_WANCONFIG=NO
>>> WAN_SCRIPTS_DIR=/etc/wanpipe/scripts
>>> WAN_FIRMWARE_DIR=/etc/wanpipe/firmware
>>> WAN_DEVICES_REV_STOP_ORDER=YES
>>> WAN_DEVICES="wanpipe1 wanpipe2 wanpipe3 "
>>>
>>> # cat /etc/wanpipe/wanpipe1.conf
>>> #================================================
>>> # WANPIPE1 Configuration File
>>> #================================================
>>> #
>>> # Note: This file was generated automatically
>>> # ? ? ? by /usr/local/sbin/setup-sangoma program.
>>> #
>>> # ? ? ? If you want to edit this file, it is
>>> # ? ? ? recommended that you use wancfg program
>>> # ? ? ? to do so.
>>> #================================================
>>> # Sangoma Technologies Inc.
>>> #================================================
>>>
>>>
>>>
>>>
>>> [devices]
>>> wanpipe1 = WAN_AFT_ISDN_BRI, Comment
>>>
>>> [interfaces]
>>> w1g1 = wanpipe1, , TDM_VOICE_API, Comment
>>>
>>> [wanpipe1]
>>> CARD_TYPE ? ? = AFT
>>> S514CPU ? ? = A
>>> CommPort ? ? = PRI
>>> AUTO_PCISLOT ? ? = NO
>>> PCISLOT ? ? = 0
>>> PCIBUS ? ? ?= 5
>>> FE_MEDIA ? ?= BRI
>>> FE_LINE ? ? ? ?= 1
>>> TDMV_LAW ? ?= ALAW
>>> RM_BRI_CLOCK_MASTER ? ?= NO
>>> MTU ? ? ? ? = 1500
>>> UDPPORT ? ? = 9000
>>> TTL ? ? ? ?= 255
>>> IGNORE_FRONT_END = NO
>>> TDMV_SPAN ? ?= 1
>>> TE_AIS_MAINTENANCE = NO ? ? ? ? ? ?#NO: defualt ?YES: Start port in AIS
>>> Blue Alarm and keep line down
>>> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?#wanpipemon -i w1g1 -c Ttx_ais_off to
>>> disable AIS maintenance mode
>>> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?#wanpipemon -i w1g1 -c Ttx_ais_on to
>>> enable AIS maintenance mode
>>> TDMV_HW_DTMF ? ? ? ?= YES ? ? ? ?# YES: receive dtmf events from hardware
>>> TDMV_HW_FAX_DETECT ? ? ?= YES ? ? # YES: receive fax 1100hz events from
>>> hardware
>>> HWEC_OPERATION_MODE ? ? = OCT_NORMAL ? ?# OCT_NORMAL: echo cancelation
>>> enabled with nlp (default)
>>> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?# OCT_SPEECH: improves software
>>> tone detection by disabling NLP (echo possible)
>>> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?# OCT_NO_ECHO:disables echo
>>> cancelation but allows VQE/tone functions.
>>> HWEC_DTMF_REMOVAL ? ? ? = NO ? ?# NO: default ?YES: remove dtmf out of
>>> incoming media (must have hwdtmf enabled)
>>> HWEC_NOISE_REDUCTION ? ?= NO ? ?# NO: default ?YES: reduces noise on the
>>> line - could break fax
>>> HWEC_ACUSTIC_ECHO ? ? ? = NO ? ?# NO: default ?YES: enables acustic echo
>>> cancelation
>>> HWEC_NLP_DISABLE ? ? ? ?= NO ? ?# NO: default ?YES: guarantees software
>>> tone detection (possible echo)
>>> HWEC_TX_AUTO_GAIN ? ? ? = 0 ? ? # 0: disable ? -40-0: default tx audio
>>> level to be maintained (-20 default)
>>> HWEC_RX_AUTO_GAIN ? ? ? = 0 ? ? # 0: disable ? -40-0: default tx audio
>>> level to be maintained (-20 default)
>>> HWEC_TX_GAIN ? ? ? ? ? ?= 0 ? ? # 0: disable ? -24-24: db values to be
>>> applied to tx signal
>>> HWEC_RX_GAIN ? ? ? ? ? ?= 0 ? ? # 0: disable ? -24-24: db values to be
>>> applied to tx signal
>>>
>>> [w1g1]
>>> ACTIVE_CH ? ?= ALL
>>> TDMV_HWEC ? ?= YES
>>> MTU ? ? ? ? = 80
>>>
>>> ? ? ... and wanpipe2 was also automatically generated so just has '2' for
>>> FE_LINE and TDMV_SPAN instead of '1'.
>>>
>>> # cat /usr/local/freeswitch/conf/freetdm.conf
>>> [span wanpipe wp1]
>>> trunk_type => ? bri
>>> group=1
>>> b-channel => ? 1:1-2
>>> d-channel => ? 1:3
>>>
>>> [span wanpipe wp2]
>>> trunk_type => ? bri
>>> group=1
>>> b-channel => ? 2:1-2
>>> d-channel => ? 2:3
>>>
>>> [span wanpipe FXS]
>>> name => ? freetdm
>>> trunk_type => ? fxs
>>> group => ? grp2
>>> fxs-channel => ? 3:1
>>>
>>> trunk_type => ? fxs
>>> group => ? grp2
>>> fxs-channel => ? 3:2
>>>
>>> # cat /usr/local/freeswitch/conf/autoload_configs/freetdm.conf.xml
>>> <configuration name="freetdm.conf" description="Freetdm Configuration">
>>> <settings>
>>> <param name="debug" value="0"/>
>>> <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>> <!--<param name="enable-analog-option" value="call-swap"/>-->
>>> <!--<param name="enable-analog-option" value="3-way"/>-->
>>> </settings>
>>>
>>> <config_profiles>
>>>
>>> <profile name="my_bri_te_1">
>>> <param name="switchtype" value="euroisdn" />
>>> <param name="interface" value="cpe"/>
>>> <param name="facility" value="no" />
>>> </profile>
>>>
>>> </config_profiles>
>>>
>>> <sangoma_bri_spans>
>>>
>>> <span name="wp1" cfgprofile="my_bri_te_1">
>>> <param name="dialplan" value="XML"/>
>>> <param name="context" value="public"/>
>>> </span>
>>>
>>> <span name="wp2" cfgprofile="my_bri_te_1">
>>> <param name="dialplan" value="XML"/>
>>> <param name="context" value="public"/>
>>> </span>
>>>
>>> </sangoma_bri_spans>
>>>
>>>
>>> <analog_spans>
>>>
>>> <span name="FXS">
>>> <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>> <param name="dialplan" value="XML"/>
>>> <param name="context" value="default"/>
>>> <!-- regex to stop dialing when it matches -->
>>> <!--<param name="dial-regex" value="5555"/>-->
>>> <!-- regex to stop dialing when it does not match -->
>>> <!--<param name="fail-dial-regex" value="^5"/>-->
>>> </span>
>>>
>>> </analog_spans>
>>>
>>> </configuration>
>>>
>>> What other config files do I need to supply? I don't want to deluge the
>>> list any more in my first post!
>>>
>>> 2. We have 2 Single Number DDI numbers configured over the 4 channels.
>>> I want to set up outgoing calls so that they can appear to come from
>>> either of the two numbers, but at present the outgoing CLI appears to be
>>> overridden by the telco (BT) to only use one of the numbers. ?Has anyone
>>> got this working in UK, and what is the format for the
>>> outbound_caller_id_number: last 6 digits or full 11 digits? ?I note that
>>> the inbound called number is only the last 6 digits.
>>>
>>> 3. I have built FreeSWITCH from git and installed at /usr/local/... and
>>> then followed the steps on the Ubuntu page in the Wiki to set up the run
>>> control scripts, etc, and run FS non-root as freeswitch:daemon. ?With
>>> FreeTDM, I have discovered that the /dev/wan* devices are owned by
>>> root:root, and so are inaccessible to FS running as non-root. ?So for
>>> now I have added a line in /etc/init.d/freeswitch to 'chgrp freeswitch
>>> /dev/wan*'. ?This is not the most elegant solution, because 'wanrouter
>>> restart' (which seems to be my best friend at the moment) resets the
>>> ownership to root:root. ?I have tried grepping to see where the mknods
>>> are for these devices, but have been unsuccessful. ?Is there a better
>>> place to 'permanently' change the device ownership?
>>>
>>> Thanks for all the great support I have already got just from editing my
>>> Wiki User page; this is a friendly group!
>>>
>>> John
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From fischetti at digiunit.it  Fri Jun 10 23:08:00 2011
From: fischetti at digiunit.it (Carmelo Fischetti - Digi Unit)
Date: Fri, 10 Jun 2011 21:08:00 +0200
Subject: [Freeswitch-users] Strange and annoying problem.
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAAE80uuUA8mNNh/+IJ5oVSWzCgAAAEAAAAM+9lscUscZPpggZgL4h3SgBAAAAAA==@digiunit.it>

Hello to everybody,
I'm new to this list but I've been experimenting with FreeSWITCH for 3
months. My environment consists of a FreeSWITCH server and some (4-5) SIP
softphones.

We are noting a strange and annoying problem. When in call, after exactly 15
minutes, FreeSWITCH CLI logs the following lines:

2011-06-10 20:10:51.689289 [DEBUG] sofia.c:4769 Channel
sofia/internal/1010 at fscch00.dyndns-server.com entering state [calling][0]
2011-06-10 20:10:51.789283 [INFO] sofia.c:748
sofia/internal/1010 at fscch00.dyndns-server.com Update Callee ID to "Outbound
Call" <1010>
2011-06-10 20:10:51.789283 [DEBUG] sofia.c:4769 Channel
sofia/internal/1010 at fscch00.dyndns-server.com entering state [ready][200]
2011-06-10 20:10:51.789283 [DEBUG] sofia.c:4780 Remote SDP:
v=0
o=amsip 0 0 IN IP4 192.168.1.100
s=talk
c=IN IP4 192.168.1.100
t=0 0
m=audio 12100 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=setup:both
a=ptime:20

2011-06-10 20:10:51.789283 [DEBUG] sofia_glue.c:4658 Audio Codec Compare
[GSM:3:8000:20:13200]/[GSM:3:8000:20:13200]
2011-06-10 20:10:51.789283 [DEBUG] sofia_glue.c:2724 Already using GSM
2011-06-10 20:10:51.789283 [DEBUG] sofia_glue.c:4772 Set 2833 dtmf send/recv
payload to 101
2011-06-10 20:10:51.789283 [DEBUG] sofia.c:5238 Processing updated SDP
2011-06-10 20:10:51.789283 [DEBUG] sofia_glue.c:3009 Audio params are
unchanged for sofia/internal/1010 at fscch00.dyndns-server.com.

My first question is: what does this mean? Is there some configuration
parameter (where?) that triggers some event after 15 minutes?

Actually, for unencrypted calls, nothing happens but when using SRTP, the
log changes to:

2011-06-10 20:32:32.825843 [DEBUG] sofia.c:4769 Channel
sofia/internal/1010 at fscch00.dyndns-server.com entering state [calling][0]
2011-06-10 20:32:32.965833 [INFO] sofia.c:748
sofia/internal/1010 at fscch00.dyndns-server.com Update Callee ID to "Outbound
Call" <1010>
2011-06-10 20:32:32.965833 [DEBUG] sofia.c:4769 Channel
sofia/internal/1010 at fscch00.dyndns-server.com entering state [ready][200]
2011-06-10 20:32:32.965833 [DEBUG] sofia.c:4780 Remote SDP:
v=0
o=amsip 0 0 IN IP4 192.168.1.100
s=talk
c=IN IP4 192.168.1.100
t=0 0
m=audio 12665 RTP/SAVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=setup:both
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:YjUwMzQyNWU4YmY2ZjhiOGYyYTQ1MmZiOGNhMWEy
a=encryption:optional
a=ptime:20

2011-06-10 20:32:32.965833 [DEBUG] sofia_glue.c:4463 Existing key is still
valid.
2011-06-10 20:32:32.965833 [DEBUG] sofia_glue.c:4658 Audio Codec Compare
[GSM:3:8000:20:13200]/[GSM:3:8000:20:13200]
2011-06-10 20:32:32.965833 [DEBUG] sofia_glue.c:2724 Already using GSM
2011-06-10 20:32:32.965833 [DEBUG] sofia_glue.c:4772 Set 2833 dtmf send/recv
payload to 101
2011-06-10 20:32:32.965833 [DEBUG] sofia.c:5238 Processing updated SDP
2011-06-10 20:32:32.965833 [DEBUG] sofia_glue.c:3009 Audio params are
unchanged for sofia/internal/1010 at fscch00.dyndns-server.com.
2011-06-10 20:32:33.485800 [ERR] switch_rtp.c:2560 Error: SRTP unprotect
failed with code 7 (auth check failed)
(the last line repeated indefinitely)...

and one leg of the call is dropped (one of the phones cannot hear the
other).
Since this happens with various softphones, it's a server issue, I suppose.

Any help would be greatly appreciated.
Thanks in advance.

Carmelo




From wes-fs at 499x.com  Sat Jun 11 00:05:13 2011
From: wes-fs at 499x.com (wes-fs at 499x.com)
Date: Fri, 10 Jun 2011 15:05:13 -0500
Subject: [Freeswitch-users] using fs api to originate a call and record it
Message-ID: <4DF278F9.6000201@499x.com>

I'm extremely new to freeswitch, and anything like it, so please forgive me!

I'd like to be able to use the API to originate a call and record it.  
So far, I have the following figured out:

originate sofia/sipinterface_1/912625551212 at 141.xxx.xxx.xxx 
&record(/tmp/myrecording.wav)
And it dials out to an external number and successfully records the 
conversation.

But, it also records the ringing.  When I was trying this through a 
dialplan, I found an option:
<action application="set" data="media_bug_answer_req=true"/>
which delayed the recording until the call was answered.

Is there a way to do the same thing through the API?

Thanks!
Wes


From msc at freeswitch.org  Sat Jun 11 02:18:49 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 10 Jun 2011 15:18:49 -0700
Subject: [Freeswitch-users] Default Install Extensions NEWBIE Question
In-Reply-To: <4DF0FE6F.2020806@gmail.com>
References: <4DF0FE6F.2020806@gmail.com>
Message-ID: <BANLkTinAcpa2cw-0mDoHVZdA2d6SDwQuHw@mail.gmail.com>

What kind of hardware, RAM, HDDs, etc? Most likely this will be a piece of
cake for FS unless you are trying to run it on a 486 with 4MB of RAM...

-MC

On Thu, Jun 9, 2011 at 10:10 AM, Mehma Sarja <mehmasarja at gmail.com> wrote:

> The default install on ubuntu renders a working system servicing 19
> extensions. Can I use this as-is in a production environment which only
> requires within-LAN, inter-departmental voice connectivity?
>
> Mehma
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Sat Jun 11 02:27:23 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 10 Jun 2011 15:27:23 -0700
Subject: [Freeswitch-users] originating a call from an outbound socket
	app
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD021F45EBBC@NY1-EXMB-01.ip-soft.net>
References: <6A6B4C284AD15042B429EB9D904544AD021F45EAA0@NY1-EXMB-01.ip-soft.net>
	<BANLkTi=LX=cGqSVrT6o-0OOPXPOA2U=MnA@mail.gmail.com>
	<6A6B4C284AD15042B429EB9D904544AD021F45EB71@NY1-EXMB-01.ip-soft.net>
	<BANLkTikG8TQ+1JhAVr8Y6xMWhH1xqor8zA@mail.gmail.com>
	<6A6B4C284AD15042B429EB9D904544AD021F45EBBC@NY1-EXMB-01.ip-soft.net>
Message-ID: <BANLkTimhqY1SuQMeWFm1mv=CvQXm72RbEw@mail.gmail.com>

On Fri, Jun 10, 2011 at 7:00 AM, Hector Geraldino <
Hector.Geraldino at ip-soft.net> wrote:

> Thanks for your response Michael.
>
>
>
> Finally I found the way of dialing to an external extension. What I did was
> just open a shell, listen in a port using nc  and tried to originate a call.
> It worked using the syntax:
>
>
>
> originate sofia/internal/5562 at 192.168.23.104 &park
>
>
>
> As this is an outbound call (not managed by the socket application) I had
> to use a new inbound socket connection to connect to FS, make the call,
> query for the uuid and receive some events (like DTMF among others).
>  Everything was going great, until I found two issues:
>
>
>
> -          The CHANNEL_ANSWER event is always triggered, no matter if the
> call is actually picked up or not on my phone. I thought that maybe I can
> use some other event to determine if the call have been answered or not
> (like CHANNEL_CALLSTATE or CALL_UPDATE), but the additional information on
> the event headers is the same no matter if it?s answered or not.
>
> -          When the call is answered, and the user presses a key to accept
> the call, the function returns the UUID of this session to the main method.
> Then I tried to make the bridge between the two calls (the legA call handled
> by the socket application, and the legB outbound call generated in the
> application using an inbound connection). I using the command
>
>
>
> api uuid_bridge <incoming_uuid> <outbound_uuid>
>
>
>
> but this didn?t work. Am I missing something here?
>
You're missing the debug log on the pastebin. ;)

Yeah, let's take a look at the debug log of this happening. Capture from the
beginning of the a leg all the way through to trying the uuid_bridge.
-MC
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From msc at freeswitch.org  Sat Jun 11 02:30:19 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 10 Jun 2011 15:30:19 -0700
Subject: [Freeswitch-users] using fs api to originate a call and record
	it
In-Reply-To: <4DF278F9.6000201@499x.com>
References: <4DF278F9.6000201@499x.com>
Message-ID: <BANLkTik6Wh7vccJrCx7rUoKSJyYcAzzr3g@mail.gmail.com>

Try this:

originate
{media_bug_answer_req=true}sofia/sipinterface_1/912625551212 at 141.xxx.xxx.xxx
 &record(/tmp/myrecording.wav)

Let us know if it works.
-MC

On Fri, Jun 10, 2011 at 1:05 PM, <wes-fs at 499x.com> wrote:

> I'm extremely new to freeswitch, and anything like it, so please forgive
> me!
>
> I'd like to be able to use the API to originate a call and record it.
> So far, I have the following figured out:
>
> originate sofia/sipinterface_1/912625551212 at 141.xxx.xxx.xxx
> &record(/tmp/myrecording.wav)
> And it dials out to an external number and successfully records the
> conversation.
>
> But, it also records the ringing.  When I was trying this through a
> dialplan, I found an option:
> <action application="set" data="media_bug_answer_req=true"/>
> which delayed the recording until the call was answered.
>
> Is there a way to do the same thing through the API?
>
> Thanks!
> Wes
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Sat Jun 11 02:46:35 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 10 Jun 2011 15:46:35 -0700
Subject: [Freeswitch-users] how to redirect $${sounds_dir} ?
In-Reply-To: <BANLkTikiLd0OWyHMz0znc14MNH79HAOzSg@mail.gmail.com>
References: <BANLkTinv+-0wP0NPyAVTRsG595HPCmUObtTNfy8m0o7f7_Gspw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DEE587AA@cooper>
	<BANLkTikiLd0OWyHMz0znc14MNH79HAOzSg@mail.gmail.com>
Message-ID: <BANLkTimfH8pR7jJb-pUxLe9b_K9WHXth4A@mail.gmail.com>

On Fri, Jun 10, 2011 at 10:40 AM, jesse <chat2jesse at gmail.com> wrote:

> vars.xml only sets sound_prefix :
>
>  <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/>
>
> $${sounds_dir} is like $${local_ip_v4}, automatically fetched by the
> application.
>
> not sure whether i can set sounds_dir in vars.xml...
>

the sounds_dir variable you cannot - it is like the base_dir, htdocs_dir,
etc. variables. What are you doing that requires you to change
${sounds_dir}? Perhaps we can help you with an alternate solution.

-MC
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From msc at freeswitch.org  Sat Jun 11 02:50:11 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 10 Jun 2011 15:50:11 -0700
Subject: [Freeswitch-users] fetch password
In-Reply-To: <BANLkTikEvezO5RH2UuEQy8cdUvZhQFouRw@mail.gmail.com>
References: <BANLkTinbM8ZW7DCyBYWp3NoMU8fU72-wQQ@mail.gmail.com>
	<BANLkTimikq6z2T+thzDm8ZDDZx7yFAhUNA@mail.gmail.com>
	<BANLkTimZv5uM8+atzR15qds8Ap3UHVbU6Q@mail.gmail.com>
	<BANLkTikEvezO5RH2UuEQy8cdUvZhQFouRw@mail.gmail.com>
Message-ID: <BANLkTi=xzS_WjE6ndcHbBu66PyGKa1Ptkg@mail.gmail.com>

If I understand you correctly you want your VM users to have their
individual passwords stored in a database. The solution is to use
mod_xml_curl instead of static XML for your directory. See mod_xml_curl on
the wiki, specifically look for the section on "bindings" for directory.

-MC

On Fri, Jun 10, 2011 at 1:29 AM, Christian Benke <benkokakao at gmail.com>wrote:

> On 10 June 2011 09:38, Sam <u2nsam at gmail.com> wrote:
> > How Does it access database , does we need to write some syntax
> somewhere,
> > so that FS fetches it out from DB.
>
> Hi!
>
> I'm still not sure if i understand your question.
>
> The default configuration uses a sqlite-DB which is saved in
> /usr/local/freeswitch/db/voicemail_default.db. The
> voicemail-application accesses this db without additional
> configuration.
>
> If you however want to save the voicemail-preferences in a
> non-embedded DB like PostgreSQL or MySQL, you have to configure odbc
> on your host to access them plus activate odbc in voicemail.conf.xml.
>
> Please read the following pages if that is what you want to do:
> http://wiki.freeswitch.org/wiki/Mod_voicemail#odbc-dsn
> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
>
> hth,
> Christian
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mehmasarja at gmail.com  Sat Jun 11 03:54:06 2011
From: mehmasarja at gmail.com (Mehma Sarja)
Date: Fri, 10 Jun 2011 16:54:06 -0700
Subject: [Freeswitch-users] Default Install Extensions NEWBIE Question
In-Reply-To: <BANLkTinAcpa2cw-0mDoHVZdA2d6SDwQuHw@mail.gmail.com>
References: <4DF0FE6F.2020806@gmail.com>
	<BANLkTinAcpa2cw-0mDoHVZdA2d6SDwQuHw@mail.gmail.com>
Message-ID: <4DF2AE9E.5040108@gmail.com>

SuperMicro Atom D510 with 4 GB RAM on Ubuntu 64 bit server OS and SSD 
drive. Need about 62 extensions. Any idea what the call quality is going 
to be like on a 100 MB client to switch and 1 GB switch to server link 
and with say, 10 concurrent conversations?

Mehma
===

On 6/10/11 3:18 PM, Michael Collins wrote:
> What kind of hardware, RAM, HDDs, etc? Most likely this will be a 
> piece of cake for FS unless you are trying to run it on a 486 with 4MB 
> of RAM...
>
> -MC
>
> On Thu, Jun 9, 2011 at 10:10 AM, Mehma Sarja <mehmasarja at gmail.com 
> <mailto:mehmasarja at gmail.com>> wrote:
>
>     The default install on ubuntu renders a working system servicing 19
>     extensions. Can I use this as-is in a production environment which
>     only
>     requires within-LAN, inter-departmental voice connectivity?
>

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From benkokakao at gmail.com  Sat Jun 11 10:47:11 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Sat, 11 Jun 2011 08:47:11 +0200
Subject: [Freeswitch-users] Default Install Extensions NEWBIE Question
In-Reply-To: <4DF2AE9E.5040108@gmail.com>
References: <4DF0FE6F.2020806@gmail.com>
	<BANLkTinAcpa2cw-0mDoHVZdA2d6SDwQuHw@mail.gmail.com>
	<4DF2AE9E.5040108@gmail.com>
Message-ID: <BANLkTinbNuR=i7ea3M5FW6mYDJ3VRVSb5Q@mail.gmail.com>

On 11 June 2011 01:54, Mehma Sarja <mehmasarja at gmail.com> wrote:
> SuperMicro Atom D510 with 4 GB RAM on Ubuntu 64 bit server OS and SSD drive.
> Need about 62 extensions. Any idea what the call quality is going to be like
> on a 100 MB client to switch and 1 GB switch to server link and with say, 10
> concurrent conversations?

I've been testing FS on an Atom N270 and started to get little audio
blemishes at around 30 concurrent conversations - so i'm pretty sure
you'll be fine(Unless all 62 extensions talk at the same time).

Your bandwith will also be sufficient - see e.g.
http://www.techrepublic.com/article/meet-voip-bandwidth-requirements-without-crippling-your-network-performance/6159446
for bandwith used by different codecs(G711 would be the most common,
G722/G722.1 "HD"-audio with better quality, if your clients support it
- see http://wiki.freeswitch.org/wiki/Codecs too).

Best regards,
Christian


From david.ponzone at ipeva.fr  Sat Jun 11 16:06:07 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sat, 11 Jun 2011 14:06:07 +0200
Subject: [Freeswitch-users] session:read() not timeouting
Message-ID: <9911DF4D-FC78-4F88-AB15-2FAB939F3A59@ipeva.fr>

All,

When using session:read() in a LUA script called from dialplan, the interdigit timeout works fine on my test box (with GIT from end of May, and a 1000hz timer).
On my prod box, the timeout does not trigger the end of the read app. The terminating character is the only way to end the read() normally.
The prod box is using a GIT from December 2010, and a bad timer (250Hz).

Should I suppose the issue comes from the timer or from the old version on the prod box (I am upgrading ASAP) ?

Thanks

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

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

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




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From roger.castaldo at gmail.com  Sat Jun 11 16:57:23 2011
From: roger.castaldo at gmail.com (Roger Castaldo)
Date: Sat, 11 Jun 2011 08:57:23 -0400
Subject: [Freeswitch-users] Default Install Extensions NEWBIE Question
In-Reply-To: <BANLkTinbNuR=i7ea3M5FW6mYDJ3VRVSb5Q@mail.gmail.com>
References: <4DF0FE6F.2020806@gmail.com>
	<BANLkTinAcpa2cw-0mDoHVZdA2d6SDwQuHw@mail.gmail.com>
	<4DF2AE9E.5040108@gmail.com>
	<BANLkTinbNuR=i7ea3M5FW6mYDJ3VRVSb5Q@mail.gmail.com>
Message-ID: <BANLkTinFYf=GmBTDdQrN=vmz3SupWRa30g@mail.gmail.com>

Also factor in the D510 is a dual core whereas the N270 is a single core.

On Sat, Jun 11, 2011 at 2:47 AM, Christian Benke <benkokakao at gmail.com>wrote:

> On 11 June 2011 01:54, Mehma Sarja <mehmasarja at gmail.com> wrote:
> > SuperMicro Atom D510 with 4 GB RAM on Ubuntu 64 bit server OS and SSD
> drive.
> > Need about 62 extensions. Any idea what the call quality is going to be
> like
> > on a 100 MB client to switch and 1 GB switch to server link and with say,
> 10
> > concurrent conversations?
>
> I've been testing FS on an Atom N270 and started to get little audio
> blemishes at around 30 concurrent conversations - so i'm pretty sure
> you'll be fine(Unless all 62 extensions talk at the same time).
>
> Your bandwith will also be sufficient - see e.g.
>
> http://www.techrepublic.com/article/meet-voip-bandwidth-requirements-without-crippling-your-network-performance/6159446
> for bandwith used by different codecs(G711 would be the most common,
> G722/G722.1 "HD"-audio with better quality, if your clients support it
> - see http://wiki.freeswitch.org/wiki/Codecs too).
>
> Best regards,
> Christian
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From yungwei at resolvity.com  Sat Jun 11 19:22:04 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Sat, 11 Jun 2011 11:22:04 -0400
Subject: [Freeswitch-users] Listen to the conversation between a user and an
	IVR
Message-ID: <33095823FD21DF429B481B5163264B7950AC643016@VMBX102.ihostexchange.net>

Hi,

I'm wondering if there's a way to listen to the conversation between a user and an IVR (javascript programs).

Here's the background info.
I have a javascript that first makes an outbound call to someone. And When the call is answered, the call is then connected another extension, which runs an IVR application written in javascript.
I want to be able to listen to the live conversation between the person and the IVR.

Is this supported? If so, how can I do that in javascript? Thanks.


From anton.vazir at gmail.com  Sat Jun 11 19:40:41 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 11 Jun 2011 20:40:41 +0500
Subject: [Freeswitch-users] ESL Inbound - what would be the best known way
 to wait in XML dialplan for any command execution via ESL?
Message-ID: <BANLkTikt2R7Qgtm6LL2jr8CP5X4_0_=aeA@mail.gmail.com>

Hi!

Switched from outbound ESL to an INBOUND mode, for controlling the
incoming CALL flow: so when call comes, Inbound ESL app catches
corresponding event and executes the necessary command on the caller
UUID.

My question is: In XML dialplan, when call comes, if do not PARK it or
do PAUSE, the ESL controlling application is too late to execute park
before XML dialplan ends, and disconnect the call.

PARK shows itself as not a good solution, since if call did not
handled by ESL - it stales.
PAUSE - i would not like to add extra delays to call handling.

I suppose there should be a known good way to handle in XML dialplan
if anything else executed on the given channel/uuid?

Regards,
Anton.


From peter.olsson at visionutveckling.se  Sat Jun 11 19:55:57 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sat, 11 Jun 2011 17:55:57 +0200
Subject: [Freeswitch-users] ESL Inbound - what would be the best known
 way to wait in XML dialplan for any command execution via ESL?
In-Reply-To: <BANLkTikt2R7Qgtm6LL2jr8CP5X4_0_=aeA@mail.gmail.com>
References: <BANLkTikt2R7Qgtm6LL2jr8CP5X4_0_=aeA@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59DEF56437@cooper>

PARK is the only way to handle this in a good way. If you can't separate the calls directly in the dial plan, you could probably do a transfer of the call (that you don't want to handle) inside your app, to another extension in the dial plan.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Anton VG [anton.vazir at gmail.com]
Skickat: den 11 juni 2011 17:40
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] ESL Inbound - what would be the best known way to wait in XML dialplan for any command execution via ESL?

Hi!

Switched from outbound ESL to an INBOUND mode, for controlling the
incoming CALL flow: so when call comes, Inbound ESL app catches
corresponding event and executes the necessary command on the caller
UUID.

My question is: In XML dialplan, when call comes, if do not PARK it or
do PAUSE, the ESL controlling application is too late to execute park
before XML dialplan ends, and disconnect the call.

PARK shows itself as not a good solution, since if call did not
handled by ESL - it stales.
PAUSE - i would not like to add extra delays to call handling.

I suppose there should be a known good way to handle in XML dialplan
if anything else executed on the given channel/uuid?

Regards,
Anton.

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

!DSPAM:4df38d5032765714154118!



From u2nsam at gmail.com  Sat Jun 11 20:04:52 2011
From: u2nsam at gmail.com (Sam)
Date: Sat, 11 Jun 2011 21:34:52 +0530
Subject: [Freeswitch-users] fetch password
In-Reply-To: <BANLkTi=xzS_WjE6ndcHbBu66PyGKa1Ptkg@mail.gmail.com>
References: <BANLkTinbM8ZW7DCyBYWp3NoMU8fU72-wQQ@mail.gmail.com>
	<BANLkTimikq6z2T+thzDm8ZDDZx7yFAhUNA@mail.gmail.com>
	<BANLkTimZv5uM8+atzR15qds8Ap3UHVbU6Q@mail.gmail.com>
	<BANLkTikEvezO5RH2UuEQy8cdUvZhQFouRw@mail.gmail.com>
	<BANLkTi=xzS_WjE6ndcHbBu66PyGKa1Ptkg@mail.gmail.com>
Message-ID: <BANLkTim3D98yYAUj8tPmqvZW_8wmSkM1yg@mail.gmail.com>

Thankyou.

On Sat, Jun 11, 2011 at 4:20 AM, Michael Collins <msc at freeswitch.org> wrote:

> If I understand you correctly you want your VM users to have their
> individual passwords stored in a database. The solution is to use
> mod_xml_curl instead of static XML for your directory. See mod_xml_curl on
> the wiki, specifically look for the section on "bindings" for directory.
>
> -MC
>
>
> On Fri, Jun 10, 2011 at 1:29 AM, Christian Benke <benkokakao at gmail.com>wrote:
>
>> On 10 June 2011 09:38, Sam <u2nsam at gmail.com> wrote:
>> > How Does it access database , does we need to write some syntax
>> somewhere,
>> > so that FS fetches it out from DB.
>>
>> Hi!
>>
>> I'm still not sure if i understand your question.
>>
>> The default configuration uses a sqlite-DB which is saved in
>> /usr/local/freeswitch/db/voicemail_default.db. The
>> voicemail-application accesses this db without additional
>> configuration.
>>
>> If you however want to save the voicemail-preferences in a
>> non-embedded DB like PostgreSQL or MySQL, you have to configure odbc
>> on your host to access them plus activate odbc in voicemail.conf.xml.
>>
>> Please read the following pages if that is what you want to do:
>> http://wiki.freeswitch.org/wiki/Mod_voicemail#odbc-dsn
>> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
>>
>> hth,
>> Christian
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anton.vazir at gmail.com  Sat Jun 11 20:07:29 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 11 Jun 2011 21:07:29 +0500
Subject: [Freeswitch-users] ESL Inbound - what would be the best known
 way to wait in XML dialplan for any command execution via ESL?
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59DEF56437@cooper>
References: <BANLkTikt2R7Qgtm6LL2jr8CP5X4_0_=aeA@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DEF56437@cooper>
Message-ID: <BANLkTi=CXOqpYtNDQYwHpz1Ds9nPSAP_ww@mail.gmail.com>

now actually I do park over ESL, as soon as I get channel_create
message, and there is only a single command in the XML dialplan: pause
1000. I was hoping there is a better approach to this.

PS. Outbound ESL showed itself as a terribly bad scaling solution if
implemented in python...

2011/6/11 Peter Olsson <peter.olsson at visionutveckling.se>:
> PARK is the only way to handle this in a good way. If you can't separate the calls directly in the dial plan, you could probably do a transfer of the call (that you don't want to handle) inside your app, to another extension in the dial plan.
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Anton VG [anton.vazir at gmail.com]
> Skickat: den 11 juni 2011 17:40
> Till: FreeSWITCH Users Help
> ?mne: [Freeswitch-users] ESL Inbound - what would be the best known way to wait in XML dialplan for any command execution via ESL?
>
> Hi!
>
> Switched from outbound ESL to an INBOUND mode, for controlling the
> incoming CALL flow: so when call comes, Inbound ESL app catches
> corresponding event and executes the necessary command on the caller
> UUID.
>
> My question is: In XML dialplan, when call comes, if do not PARK it or
> do PAUSE, the ESL controlling application is too late to execute park
> before XML dialplan ends, and disconnect the call.
>
> PARK shows itself as not a good solution, since if call did not
> handled by ESL - it stales.
> PAUSE - i would not like to add extra delays to call handling.
>
> I suppose there should be a known good way to handle in XML dialplan
> if anything else executed on the given channel/uuid?
>
> Regards,
> Anton.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> !DSPAM:4df38d5032765714154118!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From peter.olsson at visionutveckling.se  Sat Jun 11 20:16:59 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sat, 11 Jun 2011 18:16:59 +0200
Subject: [Freeswitch-users] ESL Inbound - what would be the best known
 way to wait in XML dialplan for any command execution via ESL?
In-Reply-To: <BANLkTi=CXOqpYtNDQYwHpz1Ds9nPSAP_ww@mail.gmail.com>
References: <BANLkTikt2R7Qgtm6LL2jr8CP5X4_0_=aeA@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DEF56437@cooper>,
	<BANLkTi=CXOqpYtNDQYwHpz1Ds9nPSAP_ww@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59DEF56438@cooper>

I do it like this;

First I separate calls in the dialplan, those calls with a destination to my "IVR" i will first set a variable, for instance "ivr=true", I then park the call in the dialplan. In my ESL app I wait for the CALL_PARKED event, I then check if the var is set on the call - if it is I start handling it.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Anton VG [anton.vazir at gmail.com]
Skickat: den 11 juni 2011 18:07
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] ESL Inbound - what would be the best known way to wait in XML dialplan for any command execution via ESL?

now actually I do park over ESL, as soon as I get channel_create
message, and there is only a single command in the XML dialplan: pause
1000. I was hoping there is a better approach to this.

PS. Outbound ESL showed itself as a terribly bad scaling solution if
implemented in python...

2011/6/11 Peter Olsson <peter.olsson at visionutveckling.se>:
> PARK is the only way to handle this in a good way. If you can't separate the calls directly in the dial plan, you could probably do a transfer of the call (that you don't want to handle) inside your app, to another extension in the dial plan.
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Anton VG [anton.vazir at gmail.com]
> Skickat: den 11 juni 2011 17:40
> Till: FreeSWITCH Users Help
> ?mne: [Freeswitch-users] ESL Inbound - what would be the best known way to wait in XML dialplan for any command execution via ESL?
>
> Hi!
>
> Switched from outbound ESL to an INBOUND mode, for controlling the
> incoming CALL flow: so when call comes, Inbound ESL app catches
> corresponding event and executes the necessary command on the caller
> UUID.
>
> My question is: In XML dialplan, when call comes, if do not PARK it or
> do PAUSE, the ESL controlling application is too late to execute park
> before XML dialplan ends, and disconnect the call.
>
> PARK shows itself as not a good solution, since if call did not
> handled by ESL - it stales.
> PAUSE - i would not like to add extra delays to call handling.
>
> I suppose there should be a known good way to handle in XML dialplan
> if anything else executed on the given channel/uuid?
>
> Regards,
> Anton.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

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

!DSPAM:4df392d332762225017462!



From anton.vazir at gmail.com  Sat Jun 11 20:53:32 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 11 Jun 2011 21:53:32 +0500
Subject: [Freeswitch-users] ESL Inbound - what would be the best known
 way to wait in XML dialplan for any command execution via ESL?
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59DEF56438@cooper>
References: <BANLkTikt2R7Qgtm6LL2jr8CP5X4_0_=aeA@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DEF56437@cooper>
	<BANLkTi=CXOqpYtNDQYwHpz1Ds9nPSAP_ww@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DEF56438@cooper>
Message-ID: <BANLkTin7x+bZD-p1-moSnQ+xxisCNrwcfQ@mail.gmail.com>

very similar...

2011/6/11 Peter Olsson <peter.olsson at visionutveckling.se>:
> I do it like this;
>
> First I separate calls in the dialplan, those calls with a destination to my "IVR" i will first set a variable, for instance "ivr=true", I then park the call in the dialplan. In my ESL app I wait for the CALL_PARKED event, I then check if the var is set on the call - if it is I start handling it.
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Anton VG [anton.vazir at gmail.com]
> Skickat: den 11 juni 2011 18:07
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] ESL Inbound - what would be the best known way to wait in XML dialplan for any command execution via ESL?
>
> now actually I do park over ESL, as soon as I get channel_create
> message, and there is only a single command in the XML dialplan: pause
> 1000. I was hoping there is a better approach to this.
>
> PS. Outbound ESL showed itself as a terribly bad scaling solution if
> implemented in python...
>
> 2011/6/11 Peter Olsson <peter.olsson at visionutveckling.se>:
>> PARK is the only way to handle this in a good way. If you can't separate the calls directly in the dial plan, you could probably do a transfer of the call (that you don't want to handle) inside your app, to another extension in the dial plan.
>>
>> /Peter
>> ________________________________________
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Anton VG [anton.vazir at gmail.com]
>> Skickat: den 11 juni 2011 17:40
>> Till: FreeSWITCH Users Help
>> ?mne: [Freeswitch-users] ESL Inbound - what would be the best known way to wait in XML dialplan for any command execution via ESL?
>>
>> Hi!
>>
>> Switched from outbound ESL to an INBOUND mode, for controlling the
>> incoming CALL flow: so when call comes, Inbound ESL app catches
>> corresponding event and executes the necessary command on the caller
>> UUID.
>>
>> My question is: In XML dialplan, when call comes, if do not PARK it or
>> do PAUSE, the ESL controlling application is too late to execute park
>> before XML dialplan ends, and disconnect the call.
>>
>> PARK shows itself as not a good solution, since if call did not
>> handled by ESL - it stales.
>> PAUSE - i would not like to add extra delays to call handling.
>>
>> I suppose there should be a known good way to handle in XML dialplan
>> if anything else executed on the given channel/uuid?
>>
>> Regards,
>> Anton.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> !DSPAM:4df392d332762225017462!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From anton.vazir at gmail.com  Sat Jun 11 20:55:02 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 11 Jun 2011 21:55:02 +0500
Subject: [Freeswitch-users] ESL Inbound - what would be the best known
 way to wait in XML dialplan for any command execution via ESL?
In-Reply-To: <BANLkTin7x+bZD-p1-moSnQ+xxisCNrwcfQ@mail.gmail.com>
References: <BANLkTikt2R7Qgtm6LL2jr8CP5X4_0_=aeA@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DEF56437@cooper>
	<BANLkTi=CXOqpYtNDQYwHpz1Ds9nPSAP_ww@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DEF56438@cooper>
	<BANLkTin7x+bZD-p1-moSnQ+xxisCNrwcfQ@mail.gmail.com>
Message-ID: <BANLkTikwX=bX8CTxB6hUejCJE=8JYB5dvA@mail.gmail.com>

BTW, what language do you use for your ESL app? Any info on
performance? How many CPS?

2011/6/11 Anton VG <anton.vazir at gmail.com>:
> very similar...
>
> 2011/6/11 Peter Olsson <peter.olsson at visionutveckling.se>:
>> I do it like this;
>>
>> First I separate calls in the dialplan, those calls with a destination to my "IVR" i will first set a variable, for instance "ivr=true", I then park the call in the dialplan. In my ESL app I wait for the CALL_PARKED event, I then check if the var is set on the call - if it is I start handling it.
>>
>> /Peter
>> ________________________________________
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Anton VG [anton.vazir at gmail.com]
>> Skickat: den 11 juni 2011 18:07
>> Till: FreeSWITCH Users Help
>> ?mne: Re: [Freeswitch-users] ESL Inbound - what would be the best known way to wait in XML dialplan for any command execution via ESL?
>>
>> now actually I do park over ESL, as soon as I get channel_create
>> message, and there is only a single command in the XML dialplan: pause
>> 1000. I was hoping there is a better approach to this.
>>
>> PS. Outbound ESL showed itself as a terribly bad scaling solution if
>> implemented in python...
>>
>> 2011/6/11 Peter Olsson <peter.olsson at visionutveckling.se>:
>>> PARK is the only way to handle this in a good way. If you can't separate the calls directly in the dial plan, you could probably do a transfer of the call (that you don't want to handle) inside your app, to another extension in the dial plan.
>>>
>>> /Peter
>>> ________________________________________
>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Anton VG [anton.vazir at gmail.com]
>>> Skickat: den 11 juni 2011 17:40
>>> Till: FreeSWITCH Users Help
>>> ?mne: [Freeswitch-users] ESL Inbound - what would be the best known way to wait in XML dialplan for any command execution via ESL?
>>>
>>> Hi!
>>>
>>> Switched from outbound ESL to an INBOUND mode, for controlling the
>>> incoming CALL flow: so when call comes, Inbound ESL app catches
>>> corresponding event and executes the necessary command on the caller
>>> UUID.
>>>
>>> My question is: In XML dialplan, when call comes, if do not PARK it or
>>> do PAUSE, the ESL controlling application is too late to execute park
>>> before XML dialplan ends, and disconnect the call.
>>>
>>> PARK shows itself as not a good solution, since if call did not
>>> handled by ESL - it stales.
>>> PAUSE - i would not like to add extra delays to call handling.
>>>
>>> I suppose there should be a known good way to handle in XML dialplan
>>> if anything else executed on the given channel/uuid?
>>>
>>> Regards,
>>> Anton.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> !DSPAM:4df392d332762225017462!
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>


From mehmasarja at gmail.com  Sat Jun 11 22:18:24 2011
From: mehmasarja at gmail.com (mehma sarja)
Date: Sat, 11 Jun 2011 18:18:24 +0000
Subject: [Freeswitch-users] Default Install Extensions NEWBIE Question
In-Reply-To: <BANLkTinFYf=GmBTDdQrN=vmz3SupWRa30g@mail.gmail.com>
References: <4DF0FE6F.2020806@gmail.com>
	<BANLkTinAcpa2cw-0mDoHVZdA2d6SDwQuHw@mail.gmail.com>
	<4DF2AE9E.5040108@gmail.com>
	<BANLkTinbNuR=i7ea3M5FW6mYDJ3VRVSb5Q@mail.gmail.com>
	<BANLkTinFYf=GmBTDdQrN=vmz3SupWRa30g@mail.gmail.com>
Message-ID: <BANLkTimC-KRX0H7vFxfhOmr0pNAyO8VT+g@mail.gmail.com>

Would a single core be better for this application?

On 6/11/11, Roger Castaldo <roger.castaldo at gmail.com> wrote:
> Also factor in the D510 is a dual core whereas the N270 is a single core.
>
> On Sat, Jun 11, 2011 at 2:47 AM, Christian Benke
> <benkokakao at gmail.com>wrote:
>
>> On 11 June 2011 01:54, Mehma Sarja <mehmasarja at gmail.com> wrote:
>> > SuperMicro Atom D510 with 4 GB RAM on Ubuntu 64 bit server OS and SSD
>> drive.
>> > Need about 62 extensions. Any idea what the call quality is going to be
>> like
>> > on a 100 MB client to switch and 1 GB switch to server link and with
>> > say,
>> 10
>> > concurrent conversations?
>>
>> I've been testing FS on an Atom N270 and started to get little audio
>> blemishes at around 30 concurrent conversations - so i'm pretty sure
>> you'll be fine(Unless all 62 extensions talk at the same time).
>>
>> Your bandwith will also be sufficient - see e.g.
>>
>> http://www.techrepublic.com/article/meet-voip-bandwidth-requirements-without-crippling-your-network-performance/6159446
>> for bandwith used by different codecs(G711 would be the most common,
>> G722/G722.1 "HD"-audio with better quality, if your clients support it
>> - see http://wiki.freeswitch.org/wiki/Codecs too).
>>
>> Best regards,
>> Christian
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>


From roger.castaldo at gmail.com  Sat Jun 11 22:50:19 2011
From: roger.castaldo at gmail.com (Roger Castaldo)
Date: Sat, 11 Jun 2011 14:50:19 -0400
Subject: [Freeswitch-users] Default Install Extensions NEWBIE Question
In-Reply-To: <BANLkTimC-KRX0H7vFxfhOmr0pNAyO8VT+g@mail.gmail.com>
References: <4DF0FE6F.2020806@gmail.com>
	<BANLkTinAcpa2cw-0mDoHVZdA2d6SDwQuHw@mail.gmail.com>
	<4DF2AE9E.5040108@gmail.com>
	<BANLkTinbNuR=i7ea3M5FW6mYDJ3VRVSb5Q@mail.gmail.com>
	<BANLkTinFYf=GmBTDdQrN=vmz3SupWRa30g@mail.gmail.com>
	<BANLkTimC-KRX0H7vFxfhOmr0pNAyO8VT+g@mail.gmail.com>
Message-ID: <BANLkTikCN1-DSEOBocB1-pdi5jcB5q6kBQ@mail.gmail.com>

thats a six of one half a dozen of the other, that being said, since
freeswitch operates with a threaded design as long as the OS configures
itself properly for threading, the threads can be ran in each core allowing
for more concurrent operations and performance is gained because of the
second core.

On Sat, Jun 11, 2011 at 2:18 PM, mehma sarja <mehmasarja at gmail.com> wrote:

> Would a single core be better for this application?
>
> On 6/11/11, Roger Castaldo <roger.castaldo at gmail.com> wrote:
> > Also factor in the D510 is a dual core whereas the N270 is a single core.
> >
> > On Sat, Jun 11, 2011 at 2:47 AM, Christian Benke
> > <benkokakao at gmail.com>wrote:
> >
> >> On 11 June 2011 01:54, Mehma Sarja <mehmasarja at gmail.com> wrote:
> >> > SuperMicro Atom D510 with 4 GB RAM on Ubuntu 64 bit server OS and SSD
> >> drive.
> >> > Need about 62 extensions. Any idea what the call quality is going to
> be
> >> like
> >> > on a 100 MB client to switch and 1 GB switch to server link and with
> >> > say,
> >> 10
> >> > concurrent conversations?
> >>
> >> I've been testing FS on an Atom N270 and started to get little audio
> >> blemishes at around 30 concurrent conversations - so i'm pretty sure
> >> you'll be fine(Unless all 62 extensions talk at the same time).
> >>
> >> Your bandwith will also be sufficient - see e.g.
> >>
> >>
> http://www.techrepublic.com/article/meet-voip-bandwidth-requirements-without-crippling-your-network-performance/6159446
> >> for bandwith used by different codecs(G711 would be the most common,
> >> G722/G722.1 "HD"-audio with better quality, if your clients support it
> >> - see http://wiki.freeswitch.org/wiki/Codecs too).
> >>
> >> Best regards,
> >> Christian
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jcasale at activenetwerx.com  Sat Jun 11 23:12:36 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sat, 11 Jun 2011 19:12:36 +0000
Subject: [Freeswitch-users] regex groups
Message-ID: <assp.0143d10b6f.92097A6A775D5147B1078E3F15430B9266ED53@prato.activenetwerx.local>

I wrote up a quick dialplan and intended on using $0 for the whole number, so my regex was:
     <condition field="destination_number" expression="^0\d+$">
but $0 was empty until I rewrote it:
     <condition field="destination_number" expression="^(0\d+)$">

Is this expected, I assumed not and I was not looking for $1 etc I figured $0 to
always get populated.

Thanks,
jlc


From michal.bielicki at seventhsignal.de  Sun Jun 12 00:35:33 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Sat, 11 Jun 2011 22:35:33 +0200
Subject: [Freeswitch-users] FreeTDM - Sangoma B700 - ISDN connection
	questions - UK
In-Reply-To: <BANLkTinC2xZwwWxs6cbU+8=GTKsq8uansQ@mail.gmail.com>
References: <4DED4854.6010203@earthspike.net> <4DF1CAB8.3070706@gmail.com>
	<4DF1CF66.4050700@earthspike.net>
	<BANLkTinC2xZwwWxs6cbU+8=GTKsq8uansQ@mail.gmail.com>
Message-ID: <4B655CBA-9ADD-498E-8B0A-C5594F5E634B@seventhsignal.de>

That works on centos as well

Am 10.06.2011 um 19:27 schrieb Stephen Lewis:

> Hi John
> 
> Would be interesting to hear whether you did something along these
> lines - creating /etc/udev/rules.d/80-wanpipe.rules containing
> 
> SUBSYSTEM=="wanpipe", OWNER="freeswitch", GROUP="daemon"
> SUBSYSTEM=="wp_ec", OWNER="freeswitch", GROUP="daemon"
> 
> That's what we're doing on Ubuntu 10.04.2, and it seems to work.
> 
> I assume the solution currently described in the wiki[0] doesn't
> survive a reboot?
> 
> Thanks
> 
> Stephen
> 
> [0] http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start#Using_FreeTDM_with_the_.27direct_from_git.27_method_above
> 
> On 10 June 2011 09:01, John <freeswitch at earthspike.net> wrote:
>> Nazim,
>> 
>> Late last night this got resolved; sorry I didn't post then.  We have
>> had problems with the second line, and when these were fixed last night,
>> the problem disappeared.  So now I have a nice stable FreeSWITCH with
>> the only outputs on /log 7 being SIP re-registrations.  So, you are
>> probably correct, and it was a line quality issue.
>> 
>> I've also resolved the device permissions issue with help from Sangoma
>> support, and will post on that/update the Ubuntu Wiki page when I get a
>> moment, probably after the weekend.
>> 
>> Thanks,
>> 
>> John
>> 
>> On 10/06/11 08:41, Nazim Aghabayov wrote:
>>> Hello John,
>>> 
>>> I'm not an ISDN expert, but have some experience with PRI.
>>> Have you checked the line? Sometimes bad physical interconnects makes
>>> line noisy.
>>> Channels going up and down may indicate a physical line problem or
>>> inadequate grounding (or mixed signal / common grounds). I would disable
>>> hardware echo-canellation and hw. dtmf during the troubleshooting.
>>> On PRI it's possible to trace the D channel with "wanpipemon -i w1g1
>>> -pcap -pcap_file isdn.pcap -prot ISDN -full -systime -c trd". It should
>>> be supported on BRI too, look through wanpipemon manual.
>>> 
>>> Best Regards,
>>> Nazim
>>> 
>>> 
>>> On 06/07/2011 02:36 AM, John wrote:
>>>> Hello,
>>>> 
>>>> I have just set up a FreeSWITCH box with a Sangoma B700 connected to 2 x
>>>> ISDN2e lines (each 2B+D, so 4 voice channels) in UK, so supplied by BT
>>>> Openreach.  There are a number of anomalies that I am trying to solve.
>>>> [One of these is that 1 line is dead, but that is for BT Openreach to
>>>> resolve rather than anyone on this list.]
>>>> 
>>>> I have a few questions, but as they are all related, I hope you don't
>>>> mind them in one post.  Some basics first.  The box is an Atom dual-core
>>>> with 2GB of memory and a Sangoma B700 card.  It's built with Ubuntu
>>>> 10.04.2 LTS server 64-bit, patched and up to date, and also runs dhcpd,
>>>> lighttpd and sshd.  I built the Sangoma ISDN libraries and FreeSWITCH
>>>> using the latest git versions I could ('make current' about 2 weeks
>>>> ago).  We have incoming and outgoing calls working, but some incoming
>>>> calls ring in the caller's ear, but nothing appears on the FreeSWITCH
>>>> console, and others fail.  Rebooting the server cures this.  We have had
>>>> problems with lines being disconnected and then reconnected, and it
>>>> seems that FreeTDM/wanrouter/whatever doesn't recognise previously
>>>> disconnected lines coming back into use, because a reboot finds lines
>>>> that were previously reported disconnected ('wanrouter status' or 'ftdm
>>>> list').  We have ongoing problems with one line that is 'disconnected'
>>>> (wanpipe2/wp2) but the queries I am describing below apply equally when
>>>> both lines are connected and working.
>>>> 
>>>> 1. There seems to be a lot of ISDN 'chatter' with channels going up and
>>>> down all the time even when the switch is completely idle.  Is this
>>>> normal?  Here is my /log 7 showing two of the cycles (which appear to be
>>>> about 50s apart):
>>>> 
>>>> 2011-06-06 22:01:54.992467 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:681
>>>> [SNGISDN Q921] wp1: Protocol: Data Link connection DN(4): Disconnect
>>>> initiated(263)
>>>> 2011-06-06 22:02:29.952464 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:748
>>>> [SNGISDN Q931] s1: Interface: Down(261): Dchan(285)
>>>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
>>>> [s1c1][1:1] Signalling link status changed to DOWN
>>>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
>>>> [s1c2][1:2] Signalling link status changed to DOWN
>>>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
>>>> [s1c3][1:3] Signalling link status changed to DOWN
>>>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:200
>>>> [s1c1][1:1] Setting availability rate to:5
>>>> 2011-06-06 22:02:29.952464 [DEBUG] mod_freetdm.c:2268 got clear channel
>>>> sig [SIGSTATUS_CHANGED]
>>>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:186
>>>> [s1c1][1:1] Setting availability rate to:5
>>>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:200
>>>> [s1c2][1:2] Setting availability rate to:5
>>>> 2011-06-06 22:02:29.952464 [INFO] mod_freetdm.c:2345 1:1 signalling
>>>> changed to :DOWN
>>>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:186
>>>> [s1c2][1:2] Setting availability rate to:5
>>>> 2011-06-06 22:02:29.952464 [DEBUG] mod_freetdm.c:2268 got clear channel
>>>> sig [SIGSTATUS_CHANGED]
>>>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:200
>>>> [s1c3][1:3] Setting availability rate to:5
>>>> 2011-06-06 22:02:29.952464 [INFO] mod_freetdm.c:2345 1:2 signalling
>>>> changed to :DOWN
>>>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:186
>>>> [s1c3][1:3] Setting availability rate to:5
>>>> 2011-06-06 22:02:29.952464 [DEBUG] mod_freetdm.c:2268 got clear channel
>>>> sig [SIGSTATUS_CHANGED]
>>>> 2011-06-06 22:02:29.952464 [INFO] mod_freetdm.c:2345 1:3 signalling
>>>> changed to :DOWN
>>>> 2011-06-06 22:02:29.952464 [INFO] ftmod_sangoma_isdn_stack_rcv.c:631
>>>> Received RESTART CFM (dChan:1 ces:0 type:1)
>>>> 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:1179
>>>> Receved RESTART, but Restart Indicator IE not present
>>>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:681
>>>> [SNGISDN Q921] wp1: Protocol: Data Link connection UP(3): UA frame with
>>>> F-bit = 1(258)
>>>> 2011-06-06 22:02:29.972464 [INFO] ftmod_sangoma_isdn_stack_rcv.c:631
>>>> Received RESTART CFM (dChan:1 ces:0 type:0)
>>>> 2011-06-06 22:02:29.972464 [INFO] ftmod_sangoma_isdn_stack_rcv.c:729
>>>> [SNGISDN Q931] s1: Interface: UP(260): Dchan(285)
>>>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
>>>> [s1c1][1:1] Signalling link status changed to UP
>>>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
>>>> [s1c2][1:2] Signalling link status changed to UP
>>>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
>>>> [s1c3][1:3] Signalling link status changed to UP
>>>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:200
>>>> [s1c1][1:1] Setting availability rate to:10
>>>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:186
>>>> [s1c1][1:1] Setting availability rate to:10
>>>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:200
>>>> [s1c2][1:2] Setting availability rate to:10
>>>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:186
>>>> [s1c2][1:2] Setting availability rate to:10
>>>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:200
>>>> [s1c3][1:3] Setting availability rate to:10
>>>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:186
>>>> [s1c3][1:3] Setting availability rate to:10
>>>> 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:1179
>>>> Receved RESTART, but Restart Indicator IE not present
>>>> 2011-06-06 22:02:29.972464 [DEBUG] mod_freetdm.c:2268 got clear channel
>>>> sig [SIGSTATUS_CHANGED]
>>>> 2011-06-06 22:02:29.972464 [INFO] mod_freetdm.c:2345 1:1 signalling
>>>> changed to :UP
>>>> 2011-06-06 22:02:29.972464 [DEBUG] mod_freetdm.c:2268 got clear channel
>>>> sig [SIGSTATUS_CHANGED]
>>>> 2011-06-06 22:02:29.972464 [INFO] mod_freetdm.c:2345 1:2 signalling
>>>> changed to :UP
>>>> 2011-06-06 22:02:29.972464 [DEBUG] mod_freetdm.c:2268 got clear channel
>>>> sig [SIGSTATUS_CHANGED]
>>>> 2011-06-06 22:02:29.972464 [INFO] mod_freetdm.c:2345 1:3 signalling
>>>> changed to :UP
>>>> 2011-06-06 22:03:19.912462 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:681
>>>> [SNGISDN Q921] wp1: Protocol: Data Link connection DN(4): Disconnect
>>>> initiated(263)
>>>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:681
>>>> [SNGISDN Q921] wp1: Protocol: Data Link connection UP(3): UA frame with
>>>> F-bit = 1(258)
>>>> 2011-06-06 22:03:19.932483 [INFO] ftmod_sangoma_isdn_stack_rcv.c:631
>>>> Received RESTART CFM (dChan:1 ces:0 type:0)
>>>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:1179
>>>> Receved RESTART, but Restart Indicator IE not present
>>>> 2011-06-06 22:03:19.932483 [INFO] ftmod_sangoma_isdn_stack_rcv.c:729
>>>> [SNGISDN Q931] s1: Interface: UP(260): Dchan(285)
>>>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
>>>> [s1c1][1:1] Signalling link status changed to UP
>>>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
>>>> [s1c2][1:2] Signalling link status changed to UP
>>>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43
>>>> [s1c3][1:3] Signalling link status changed to UP
>>>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:200
>>>> [s1c1][1:1] Setting availability rate to:10
>>>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:186
>>>> [s1c1][1:1] Setting availability rate to:10
>>>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:200
>>>> [s1c2][1:2] Setting availability rate to:10
>>>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:186
>>>> [s1c2][1:2] Setting availability rate to:10
>>>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:200
>>>> [s1c3][1:3] Setting availability rate to:10
>>>> 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:186
>>>> [s1c3][1:3] Setting availability rate to:10
>>>> 2011-06-06 22:03:19.932483 [DEBUG] mod_freetdm.c:2268 got clear channel
>>>> sig [SIGSTATUS_CHANGED]
>>>> 2011-06-06 22:03:19.932483 [INFO] mod_freetdm.c:2345 1:1 signalling
>>>> changed to :UP
>>>> 2011-06-06 22:03:19.932483 [DEBUG] mod_freetdm.c:2268 got clear channel
>>>> sig [SIGSTATUS_CHANGED]
>>>> 2011-06-06 22:03:19.932483 [INFO] mod_freetdm.c:2345 1:2 signalling
>>>> changed to :UP
>>>> 2011-06-06 22:03:19.932483 [DEBUG] mod_freetdm.c:2268 got clear channel
>>>> sig [SIGSTATUS_CHANGED]
>>>> 2011-06-06 22:03:19.932483 [INFO] mod_freetdm.c:2345 1:3 signalling
>>>> changed to :UP
>>>> freeswitch at internal>   ftdm list
>>>> +OK
>>>> span: 1 (wp1)
>>>> type: Sangoma (ISDN)
>>>> physical_status: ok
>>>> signaling_status: UP
>>>> chan_count: 3
>>>> dialplan: XML
>>>> context: public
>>>> dial_regex:
>>>> fail_dial_regex:
>>>> hold_music:
>>>> analog_options: none
>>>> +OK
>>>> span: 2 (wp2)
>>>> type: Sangoma (ISDN)
>>>> physical_status: alarmed
>>>> signaling_status: DOWN
>>>> chan_count: 3
>>>> dialplan: XML
>>>> context: public
>>>> dial_regex:
>>>> fail_dial_regex:
>>>> hold_music:
>>>> analog_options: none
>>>> +OK
>>>> span: 3 (FXS)
>>>> type: analog
>>>> physical_status: ok
>>>> signaling_status: UP
>>>> chan_count: 2
>>>> dialplan: XML
>>>> context: default
>>>> dial_regex:
>>>> fail_dial_regex:
>>>> hold_music:
>>>> analog_options: none
>>>> 
>>>> 
>>>> # wanrouter status
>>>> 
>>>> Devices currently active:
>>>>        wanpipe1 wanpipe2 wanpipe3
>>>> 
>>>> 
>>>> Wanpipe Config:
>>>> 
>>>> Device name | Protocol Map | Adapter  | IRQ | Slot/IO | If's | CLK |
>>>> Baud rate |
>>>> wanpipe1    | N/A          | A500/B700| 20  | 0       | 1    | N/A |
>>>> 0         |
>>>> wanpipe2    | N/A          | A500/B700| 20  | 0       | 1    | N/A |
>>>> 0         |
>>>> wanpipe3    | N/A          | A200/A400/B600/B700/B800| 20  | 0       |
>>>> 1    | N/A | 0         |
>>>> 
>>>> Wanrouter Status:
>>>> 
>>>> Device name | Protocol | Station | Status        |
>>>> wanpipe1    | AFT ISDN | N/A     | Connected     |
>>>> wanpipe2    | AFT ISDN | N/A     | Disconnected  |
>>>> wanpipe3    | A-ANALOG | N/A     | Connected     |
>>>> 
>>>> 
>>>> # cat /etc/wanpipe/wanrouter.rc
>>>> #!/bin/sh
>>>> # .. comments snipped ...
>>>> ROUTER_BOOT=YES
>>>> WAN_CONF_DIR=/etc/wanpipe
>>>> WAN_INTR_DIR=/etc/wanpipe/interfaces
>>>> WAN_BIN_DIR=/usr/sbin
>>>> WAN_LOG=/var/log/wanrouter
>>>> WAN_LOCK=/var/lock/wanrouter
>>>> WAN_LOCK_DIR=/var/lock
>>>> WAN_IP_FORWARD=NO
>>>> NEW_IF_TYPE=NO
>>>> WAN_LIB_DIR=/etc/wanpipe/lib
>>>> WAN_ADSL_LIST=/etc/wanpipe/wan_adsl.list
>>>> WAN_ANNEXG_LOAD=NO
>>>> WAN_SCTP_LOAD=NO
>>>> WAN_LIP_LOAD=NO
>>>> WAN_DYN_WANCONFIG=NO
>>>> WAN_SCRIPTS_DIR=/etc/wanpipe/scripts
>>>> WAN_FIRMWARE_DIR=/etc/wanpipe/firmware
>>>> WAN_DEVICES_REV_STOP_ORDER=YES
>>>> WAN_DEVICES="wanpipe1 wanpipe2 wanpipe3 "
>>>> 
>>>> # cat /etc/wanpipe/wanpipe1.conf
>>>> #================================================
>>>> # WANPIPE1 Configuration File
>>>> #================================================
>>>> #
>>>> # Note: This file was generated automatically
>>>> #       by /usr/local/sbin/setup-sangoma program.
>>>> #
>>>> #       If you want to edit this file, it is
>>>> #       recommended that you use wancfg program
>>>> #       to do so.
>>>> #================================================
>>>> # Sangoma Technologies Inc.
>>>> #================================================
>>>> 
>>>> 
>>>> 
>>>> 
>>>> [devices]
>>>> wanpipe1 = WAN_AFT_ISDN_BRI, Comment
>>>> 
>>>> [interfaces]
>>>> w1g1 = wanpipe1, , TDM_VOICE_API, Comment
>>>> 
>>>> [wanpipe1]
>>>> CARD_TYPE     = AFT
>>>> S514CPU     = A
>>>> CommPort     = PRI
>>>> AUTO_PCISLOT     = NO
>>>> PCISLOT     = 0
>>>> PCIBUS      = 5
>>>> FE_MEDIA    = BRI
>>>> FE_LINE        = 1
>>>> TDMV_LAW    = ALAW
>>>> RM_BRI_CLOCK_MASTER    = NO
>>>> MTU         = 1500
>>>> UDPPORT     = 9000
>>>> TTL        = 255
>>>> IGNORE_FRONT_END = NO
>>>> TDMV_SPAN    = 1
>>>> TE_AIS_MAINTENANCE = NO            #NO: defualt  YES: Start port in AIS
>>>> Blue Alarm and keep line down
>>>>                                    #wanpipemon -i w1g1 -c Ttx_ais_off to
>>>> disable AIS maintenance mode
>>>>                                    #wanpipemon -i w1g1 -c Ttx_ais_on to
>>>> enable AIS maintenance mode
>>>> TDMV_HW_DTMF        = YES        # YES: receive dtmf events from hardware
>>>> TDMV_HW_FAX_DETECT      = YES     # YES: receive fax 1100hz events from
>>>> hardware
>>>> HWEC_OPERATION_MODE     = OCT_NORMAL    # OCT_NORMAL: echo cancelation
>>>> enabled with nlp (default)
>>>>                                            # OCT_SPEECH: improves software
>>>> tone detection by disabling NLP (echo possible)
>>>>                                            # OCT_NO_ECHO:disables echo
>>>> cancelation but allows VQE/tone functions.
>>>> HWEC_DTMF_REMOVAL       = NO    # NO: default  YES: remove dtmf out of
>>>> incoming media (must have hwdtmf enabled)
>>>> HWEC_NOISE_REDUCTION    = NO    # NO: default  YES: reduces noise on the
>>>> line - could break fax
>>>> HWEC_ACUSTIC_ECHO       = NO    # NO: default  YES: enables acustic echo
>>>> cancelation
>>>> HWEC_NLP_DISABLE        = NO    # NO: default  YES: guarantees software
>>>> tone detection (possible echo)
>>>> HWEC_TX_AUTO_GAIN       = 0     # 0: disable   -40-0: default tx audio
>>>> level to be maintained (-20 default)
>>>> HWEC_RX_AUTO_GAIN       = 0     # 0: disable   -40-0: default tx audio
>>>> level to be maintained (-20 default)
>>>> HWEC_TX_GAIN            = 0     # 0: disable   -24-24: db values to be
>>>> applied to tx signal
>>>> HWEC_RX_GAIN            = 0     # 0: disable   -24-24: db values to be
>>>> applied to tx signal
>>>> 
>>>> [w1g1]
>>>> ACTIVE_CH    = ALL
>>>> TDMV_HWEC    = YES
>>>> MTU         = 80
>>>> 
>>>>     ... and wanpipe2 was also automatically generated so just has '2' for
>>>> FE_LINE and TDMV_SPAN instead of '1'.
>>>> 
>>>> # cat /usr/local/freeswitch/conf/freetdm.conf
>>>> [span wanpipe wp1]
>>>> trunk_type =>   bri
>>>> group=1
>>>> b-channel =>   1:1-2
>>>> d-channel =>   1:3
>>>> 
>>>> [span wanpipe wp2]
>>>> trunk_type =>   bri
>>>> group=1
>>>> b-channel =>   2:1-2
>>>> d-channel =>   2:3
>>>> 
>>>> [span wanpipe FXS]
>>>> name =>   freetdm
>>>> trunk_type =>   fxs
>>>> group =>   grp2
>>>> fxs-channel =>   3:1
>>>> 
>>>> trunk_type =>   fxs
>>>> group =>   grp2
>>>> fxs-channel =>   3:2
>>>> 
>>>> # cat /usr/local/freeswitch/conf/autoload_configs/freetdm.conf.xml
>>>> <configuration name="freetdm.conf" description="Freetdm Configuration">
>>>> <settings>
>>>> <param name="debug" value="0"/>
>>>> <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>> <!--<param name="enable-analog-option" value="call-swap"/>-->
>>>> <!--<param name="enable-analog-option" value="3-way"/>-->
>>>> </settings>
>>>> 
>>>> <config_profiles>
>>>> 
>>>> <profile name="my_bri_te_1">
>>>> <param name="switchtype" value="euroisdn" />
>>>> <param name="interface" value="cpe"/>
>>>> <param name="facility" value="no" />
>>>> </profile>
>>>> 
>>>> </config_profiles>
>>>> 
>>>> <sangoma_bri_spans>
>>>> 
>>>> <span name="wp1" cfgprofile="my_bri_te_1">
>>>> <param name="dialplan" value="XML"/>
>>>> <param name="context" value="public"/>
>>>> </span>
>>>> 
>>>> <span name="wp2" cfgprofile="my_bri_te_1">
>>>> <param name="dialplan" value="XML"/>
>>>> <param name="context" value="public"/>
>>>> </span>
>>>> 
>>>> </sangoma_bri_spans>
>>>> 
>>>> 
>>>> <analog_spans>
>>>> 
>>>> <span name="FXS">
>>>> <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>> <param name="dialplan" value="XML"/>
>>>> <param name="context" value="default"/>
>>>> <!-- regex to stop dialing when it matches -->
>>>> <!--<param name="dial-regex" value="5555"/>-->
>>>> <!-- regex to stop dialing when it does not match -->
>>>> <!--<param name="fail-dial-regex" value="^5"/>-->
>>>> </span>
>>>> 
>>>> </analog_spans>
>>>> 
>>>> </configuration>
>>>> 
>>>> What other config files do I need to supply? I don't want to deluge the
>>>> list any more in my first post!
>>>> 
>>>> 2. We have 2 Single Number DDI numbers configured over the 4 channels.
>>>> I want to set up outgoing calls so that they can appear to come from
>>>> either of the two numbers, but at present the outgoing CLI appears to be
>>>> overridden by the telco (BT) to only use one of the numbers.  Has anyone
>>>> got this working in UK, and what is the format for the
>>>> outbound_caller_id_number: last 6 digits or full 11 digits?  I note that
>>>> the inbound called number is only the last 6 digits.
>>>> 
>>>> 3. I have built FreeSWITCH from git and installed at /usr/local/... and
>>>> then followed the steps on the Ubuntu page in the Wiki to set up the run
>>>> control scripts, etc, and run FS non-root as freeswitch:daemon.  With
>>>> FreeTDM, I have discovered that the /dev/wan* devices are owned by
>>>> root:root, and so are inaccessible to FS running as non-root.  So for
>>>> now I have added a line in /etc/init.d/freeswitch to 'chgrp freeswitch
>>>> /dev/wan*'.  This is not the most elegant solution, because 'wanrouter
>>>> restart' (which seems to be my best friend at the moment) resets the
>>>> ownership to root:root.  I have tried grepping to see where the mknods
>>>> are for these devices, but have been unsuccessful.  Is there a better
>>>> place to 'permanently' change the device ownership?
>>>> 
>>>> Thanks for all the great support I have already got just from editing my
>>>> Wiki User page; this is a friendly group!
>>>> 
>>>> John
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Gesch?ftsf?hrer / CEO

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

Amtsgericht Charlottenburg HRA 44413 B
Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de

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From avi at avimarcus.net  Sun Jun 12 01:20:45 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 12 Jun 2011 00:20:45 +0300
Subject: [Freeswitch-users] IVR -> Phrase with a parameter
Message-ID: <BANLkTi=LMS1jGG0E_T47Ph9TXc-8RhwwGA@mail.gmail.com>

I'm kinda new to IVRs and Phrases, but I see this:

    <action application="phrase" data="msgcount,130"/>
    <macro name="msgcount">
         <input pattern="(.*)">

Meaning I can pass a parameter to the phrase, and then set different
matching input configurations.

Now, I see phrases are being used in the IVR:
I'd like to have an announcement of a balance at the beginning, but
start listening for the digits already.
But, I'd like to have two different types of announcement in the beginning.
I can either duplicate and create two different IVRs and two different
sets of phrases, or I can take advantage of that parameter passing.

But when I try:   greet-long="phrase:cc_ivr_main_menu,${cash}" I get:
2011-06-12 00:05:24.788910 [ERR] switch_ivr_play_say.c:142 Can't find
macro cc_ivr_main_menu,33.09.

Am I missing something?
Do I need to make a lua IVR to start capturing digits while I play
other audio (how??) or create split the IVR into two nearly duplicate
ones?

Thanks,
Avi


From georuse at gmx.com  Sat Jun 11 21:08:58 2011
From: georuse at gmx.com (george russell)
Date: Sat, 11 Jun 2011 17:08:58 +0000
Subject: [Freeswitch-users] CHANNEL_BRIDGE event missing
Message-ID: <20110611170858.278240@gmx.com>

Freeswitch has connected my call. The call has gone through the gateway and connected to the phone on the other end.
 Because the call has connected I thought CHANNEL_BRIDGE should of occured.

 This is the list of events captured. Why would CHANNEL_BRIDGE event be missing even though the call has been bridged?


 I have done a "events plain all" and filtered for aleg uuid and bleg uuid

 "filter Unique-ID (aleg uuid)"
 "filter Unique-ID (bleg uuid)"



 CHANNEL_EXECUTE / export
 CHANNEL_EXECUTE_COMPLETE / export
 CHANNEL_EXECUTE / export
 CHANNEL_EXECUTE_COMPLETE / export
 CHANNEL_EXECUTE / export
 CHANNEL_EXECUTE_COMPLETE / export
 CHANNEL_EXECUTE / export
 CHANNEL_EXECUTE_COMPLETE / export
 CHANNEL_EXECUTE / export
 CHANNEL_EXECUTE_COMPLETE / export
 CHANNEL_EXECUTE / export
 CHANNEL_EXECUTE_COMPLETE / export
 CHANNEL_EXECUTE / export
 CHANNEL_EXECUTE_COMPLETE / export
 CHANNEL_EXECUTE / bridge
 CHANNEL_PROGRESS /
 CALL_UPDATE /
 CALL_UPDATE /
 CALL_UPDATE /
 CALL_UPDATE /
 CHANNEL_ANSWER /
 SESSION_HEARTBEAT /
 CHANNEL_HANGUP /
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From snabb at epipe.com  Sat Jun 11 13:02:12 2011
From: snabb at epipe.com (Janne Snabb)
Date: Sat, 11 Jun 2011 09:02:12 +0000
Subject: [Freeswitch-users] Free ZRTP implementation?
Message-ID: <alpine.BSF.2.00.1106110853570.67270@tiktik.epipe.com>

Hi,

I wanted to compile FreeSWITCH with ZRTP support, but it looks like
Zimmermann's commercial libzrtp has not been available for "evaluation"
download since the beginning of this year. I found a couple of older
messages asking about it in the mailing list archives, but no real
solution.

Does anyone have patches or is anyone working on a module which
would use a free ZRTP implementation such as ZORG or GNU ZRTP?

If not, I might be able to have a look at it at some point... no
promises though.

--
Janne Snabb / EPIPE Communications
snabb at epipe.com - http://epipe.com/



From msc at freeswitch.org  Sun Jun 12 02:07:03 2011
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 11 Jun 2011 15:07:03 -0700
Subject: [Freeswitch-users] FreeSWITCH Cookbook Need: Fax Examples
Message-ID: <BANLkTimzmAwyedYhBFuSy1TMgaQ45tZ-Xg@mail.gmail.com>

Hey all,

I am working on the FSCB and I could use some of your real-world fax
examples. I'm looking for anything like these:

Inbound fax to email/PDF
Outbound fax examples (scripts, etc. - whatever you do to make it easier for
your users to send faxes thru FS)
T.38 config examples and reports on symptoms/issues you had to overcome to
make it work

I will gladly accept any information you have, even if it's just raw dumps
of scripts, XML configs, etc. Shoot them to me off-list.

Thanks!
-MC
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From msc at freeswitch.org  Sun Jun 12 02:14:33 2011
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 11 Jun 2011 15:14:33 -0700
Subject: [Freeswitch-users] IVR -> Phrase with a parameter
In-Reply-To: <BANLkTi=LMS1jGG0E_T47Ph9TXc-8RhwwGA@mail.gmail.com>
References: <BANLkTi=LMS1jGG0E_T47Ph9TXc-8RhwwGA@mail.gmail.com>
Message-ID: <BANLkTi=j2D9Gn93g8qjqzVzvxW_GDvnG0g@mail.gmail.com>

If you had read chapter 6 of the bridge book you'd know the answer. ;)

If you want to pass arguments then do something like this:

greet-long="phrase:cc_ivr_main_menu:${cash}:${arg2}:${arg3}"

and

<macro name="cc_ivr_main_menu">
        <input pattern="(.*?):(.*?):(.*?)">
            <!-- ${cash} is in $1 now -->
            <!-- ${arg2} is in $2 now -->
            <!-- ${arg3} is in $3 now -->

Now you can use $1, $2, $3, etc. in your phrase macro. If you want to see
macros in action then I suggest you look at the voicemail macros - they do
all sorts of stuff. Of course, if you'd read chapter 6 of the FS book you'd
know that, too. ;)

-MC

P.S. - Did I mention chapter 6 of the book? :D The reason I'm so familiar
with it is because I wrote the stuff on phrase macros in chapter 6 and
chapter 7...

On Sat, Jun 11, 2011 at 2:20 PM, Avi Marcus <avi at avimarcus.net> wrote:

> I'm kinda new to IVRs and Phrases, but I see this:
>
>    <action application="phrase" data="msgcount,130"/>
>    <macro name="msgcount">
>         <input pattern="(.*)">
>
> Meaning I can pass a parameter to the phrase, and then set different
> matching input configurations.
>
> Now, I see phrases are being used in the IVR:
> I'd like to have an announcement of a balance at the beginning, but
> start listening for the digits already.
> But, I'd like to have two different types of announcement in the beginning.
> I can either duplicate and create two different IVRs and two different
> sets of phrases, or I can take advantage of that parameter passing.
>
> But when I try:   greet-long="phrase:cc_ivr_main_menu,${cash}" I get:
> 2011-06-12 00:05:24.788910 [ERR] switch_ivr_play_say.c:142 Can't find
> macro cc_ivr_main_menu,33.09.
>
> Am I missing something?
> Do I need to make a lua IVR to start capturing digits while I play
> other audio (how??) or create split the IVR into two nearly duplicate
> ones?
>
> Thanks,
> Avi
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From david.ponzone at ipeva.fr  Sun Jun 12 03:35:28 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sun, 12 Jun 2011 01:35:28 +0200
Subject: [Freeswitch-users] g729 licenses are not valid anymore after upgrade
Message-ID: <9335B0A6-5190-4B14-86FE-EE515AF0B97A@ipeva.fr>

I just upgraded from a 6 months GIT version to current, and the latest mod_com_g729 does not seem to accept my G729 licenses anymore.

The error message is:
Unrecognized resource G729A/0

If I revert back to the previous version (with mod_com_g729 158), it works fine.

Is there something that was changed in the licenses ?

Thanks

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

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

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




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From woodydickson at gmail.com  Sun Jun 12 14:13:08 2011
From: woodydickson at gmail.com (Woody Dickson)
Date: Sun, 12 Jun 2011 10:13:08 +0000 (UTC)
Subject: [Freeswitch-users] Invitation to connect on LinkedIn
Message-ID: <1517215760.4106057.1307873588760.JavaMail.app@ela4-bed78.prod>

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

   
FreeSWITCH,

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

- Woody

Woody Dickson
Managing Director at Power Telecom 
San Francisco Bay Area

Confirm that you know Woody Dickson
https://www.linkedin.com/e/xbphn8-gotu5zpw-6s/isd/3186430560/QOqMJZue/


 
-- 
(c) 2011, LinkedIn Corporation
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From devel at omninet.eu  Sun Jun 12 16:18:38 2011
From: devel at omninet.eu (Anestis Mavro)
Date: Sun, 12 Jun 2011 15:18:38 +0300
Subject: [Freeswitch-users] g729 licenses are not valid anymore after
	upgrade
In-Reply-To: <9335B0A6-5190-4B14-86FE-EE515AF0B97A@ipeva.fr>
References: <9335B0A6-5190-4B14-86FE-EE515AF0B97A@ipeva.fr>
Message-ID: <37601782DCD4427186EF316CA19FD65A@omni1.local>

Just activate the license again. I?ve had the same problem a few weeks ago


 

Regards

Anestis

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David
Ponzone
Sent: Sunday, June 12, 2011 2:35 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] g729 licenses are not valid anymore after
upgrade

 

I just upgraded from a 6 months GIT version to current, and the latest
mod_com_g729 does not seem to accept my G729 licenses anymore.

 

The error message is:

Unrecognized resource G729A/0

 

If I revert back to the previous version (with mod_com_g729 158), it works
fine.

 

Is there something that was changed in the licenses ?

 

Thanks

 

David Ponzone  Direction Technique

email: david.ponzone at ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

 

Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>   -   www.ipeva-studio.com
<BLOCKED::http://www.ipeva-studio.com/> 

 

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

 





 



__________ Information from ESET NOD32 Antivirus, version of virus signature
database 5054 (20100423) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From gustavo.espeche at easyipcall.com  Sun Jun 12 19:01:06 2011
From: gustavo.espeche at easyipcall.com (Gustavo Espeche)
Date: Sun, 12 Jun 2011 12:01:06 -0300
Subject: [Freeswitch-users] Codec/CDR Error
Message-ID: <1307890866.2056.31.camel@gustavo-laptop>

Hi we found an error in freeswitch

2011-06-12 10:45:35.662591 [ERR] sofia.c:5323 Codec Error! v=0
o=Argentina-2 1307890456 1307890458 IN IP4 200.49.30.68
s=sip call
c=IN IP4 200.49.30.69
t=0 0
m=audio 42410 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
--------------------------------------------
2011-06-12 10:48:28.322557 [ERR] sofia.c:5323 Codec Error! v=0
o=Sonus_UAC 19969 23951 IN IP4 38.105.229.116
s=SIP Media Capabilities
c=IN IP4 38.105.229.114
t=0 0
m=audio 14578 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
--------------------------------------------
2011-06-12 10:50:14.686620 [ERR] sofia.c:5323 Codec Error! v=0
o=ATSI-B2 188 2 IN IP4 208.37.200.235
s=sip call
c=IN IP4 208.37.200.245
t=0 0
m=audio 35538 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--------------------------------------------------------
2011-06-12 10:47:52.362623 [ERR] sofia.c:5323 Codec Error! v=0
o=VoipSwitch 10828 10828 IN IP4 50.30.35.13
s=VoipSIP
i=Audio Session
c=IN IP4 50.30.35.13
t=0 0
m=audio 9828 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-1

the problem is that when this error happens FS send the cdr immediately
but the call still connected, and the provider still with all billing,
because of the we see a billing difference with our provider.
The error happens with different provider or customer equipment.
Some one know if we can configure of freeswitch in some way that the
call hangup.
The problem isn't fix the codec error, for us the real problem is that
the FS do a cdr but the call still connected.
Thank a lot in advice.


-- 

Gustavo Espeche
EasyIpCall S.R.L.
www.easyipcall.com
Bv Mitre 517 24? E
Cordoba - Argentina
Te: +54 - 351 - 4280633




From nsirugudi at gmail.com  Sun Jun 12 19:32:53 2011
From: nsirugudi at gmail.com (Narendra Sirugudi)
Date: Sun, 12 Jun 2011 21:02:53 +0530
Subject: [Freeswitch-users] Originating a call from a confeence is not
	working ?
Message-ID: <BANLkTi=_ogA9pig3HTQCoG2mWB9P_a_8ng@mail.gmail.com>

Hi,

I am trying to use the dial-out feature of the conference application of
freeswitch.
I am using freeswtich version 1.0.6.

As i understand the conference gets created when the first user joins. Hence
i made one user with number 1004 join the conference.

freeswitch at internal> conference list
Conference 3001-10.10.10.16 (1 member rate: 8000)
6;sofia/internal/1004 at 10.10.10.16
;08178796-a06e-4ee8-81fb-1f0cc53f8fb5;1004;1004;hear|speak|floor;0;0;300

Now i dail out another user 1005 using the command :
freeswitch at internal> conference 3001-10.10.10.16 dial
{originate_timeout=30}sofia/internal/1005 at 10.10.10.16<%7Boriginate_timeout=30%7Dsofia/internal/1005 at 135.254.210.166>1234567890
FreeSWITCH_Conference
This does not work. I observe the following errors in the fs_cli logs:

*2011-06-12 10:57:57.621014 [ERR] sofia.c:5366 Cannot Blind Transfer 1
Legged calls*
can anyone tell what could be going wrong ?

thanks
--naren

The complete logs are given below:

#############################################################
freeswitch at internal> conference 3001-10.10.10.16 dial
{originate_timeout=30}sofia/internal/1005 at 10.10.10.16 1234567890
FreeSWITCH_Conference
Call Requested: result: [NO_USER_RESPONSE]
2011-06-12 10:57:57.606002 [DEBUG] switch_ivr_originate.c:1885 variable
string 0 = [ignore_early_media=true]
freeswitch at internal> 2011-06-12 10:57:57.606002 [DEBUG]
switch_ivr_originate.c:1885 variable string 1 = [originate_timeout=30]
2011-06-12 10:57:57.606002 [NOTICE] switch_channel.c:669 New Channel
sofia/internal/1005 at 10.10.10.16 [a77137ad-ad1a-45e9-ac27-88a996cd865d]
2011-06-12 10:57:57.606002 [DEBUG] mod_sofia.c:3384 (
sofia/internal/1005 at 10.10.10.16) State Change CS_NEW -> CS_INIT
2011-06-12 10:57:57.606002 [DEBUG] switch_core_session.c:1021 Send signal
sofia/internal/1005 at 10.10.10.16 [BREAK]
2011-06-12 10:57:57.607036 [DEBUG] switch_core_state_machine.c:314 (
sofia/internal/1005 at 10.10.10.16) Running State Change CS_INIT
2011-06-12 10:57:57.607036 [DEBUG] switch_core_state_machine.c:338 (
sofia/internal/1005 at 10.10.10.16) State INIT
2011-06-12 10:57:57.607036 [DEBUG] mod_sofia.c:83
sofia/internal/1005 at 10.10.10.16 SOFIA INIT
2011-06-12 10:57:57.607036 [DEBUG] mod_sofia.c:117 (
sofia/internal/1005 at 10.10.10.16) State Change CS_INIT -> CS_ROUTING
2011-06-12 10:57:57.607036 [DEBUG] switch_core_session.c:1021 Send signal
sofia/internal/1005 at 10.10.10.16 [BREAK]
2011-06-12 10:57:57.607036 [DEBUG] switch_core_state_machine.c:338 (
sofia/internal/1005 at 10.10.10.16) State INIT going to sleep
2011-06-12 10:57:57.607036 [DEBUG] switch_core_state_machine.c:314 (
sofia/internal/1005 at 10.10.10.16) Running State Change CS_ROUTING
2011-06-12 10:57:57.607036 [DEBUG] switch_core_state_machine.c:341 (
sofia/internal/1005 at 10.10.10.16) State ROUTING
2011-06-12 10:57:57.607036 [DEBUG] mod_sofia.c:140
sofia/internal/1005 at 10.10.10.16 SOFIA ROUTING
2011-06-12 10:57:57.607036 [DEBUG] switch_ivr_originate.c:66 (
sofia/internal/1005 at 10.10.10.16) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2011-06-12 10:57:57.607036 [DEBUG] switch_core_session.c:1021 Send signal
sofia/internal/1005 at 10.10.10.16 [BREAK]
2011-06-12 10:57:57.607036 [DEBUG] switch_core_state_machine.c:341 (
sofia/internal/1005 at 10.10.10.16) State ROUTING going to sleep
2011-06-12 10:57:57.607036 [DEBUG] switch_core_state_machine.c:314 (
sofia/internal/1005 at 10.10.10.16) Running State Change CS_CONSUME_MEDIA
2011-06-12 10:57:57.607036 [DEBUG] switch_core_state_machine.c:360 (
sofia/internal/1005 at 10.10.10.16) State CONSUME_MEDIA
2011-06-12 10:57:57.607036 [DEBUG] switch_core_state_machine.c:360 (
sofia/internal/1005 at 10.10.10.16) State CONSUME_MEDIA going to sleep
2011-06-12 10:57:57.608051 [DEBUG] sofia.c:4153 Channel
sofia/internal/1005 at 10.10.10.16 entering state [calling][0]
2011-06-12 10:57:57.608051 [DEBUG] sofia.c:5847 IP 10.10.10.16 Rejected by
acl "domains". Falling back to Digest auth.
2011-06-12 10:57:57.608051 [NOTICE] switch_channel.c:669 New Channel
sofia/internal/1234567890 at 10.10.10.16 [86e5b4a9-8c63-4d22-b087-94c4b95d1abb]
2011-06-12 10:57:57.609139 [DEBUG] switch_core_state_machine.c:314 (
sofia/internal/1234567890 at 10.10.10.16) Running State Change CS_NEW
2011-06-12 10:57:57.609139 [DEBUG] switch_core_state_machine.c:320 (
sofia/internal/1234567890 at 10.10.10.16) State NEW
2011-06-12 10:57:57.616761 [DEBUG] sofia.c:4153 Channel
sofia/internal/1234567890 at 10.10.10.16 entering state [received][100]
2011-06-12 10:57:57.616761 [DEBUG] sofia.c:4164 Remote SDP:
v=0
o=FreeSWITCH 1307870225 1307870226 IN IP4 10.10.10.16
s=FreeSWITCH
c=IN IP4 10.10.10.16
t=0 0
m=audio 20452 RTP/AVP 115 107 9 0 8 3 101 13
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:107 G7221/16000
a=fmtp:107 bitrate=32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20
2011-06-12 10:57:57.616761 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
[G7221:115:32000:20]/[G7221:115:32000:20]
2011-06-12 10:57:57.616761 [DEBUG] sofia_glue.c:2354 Set Codec
sofia/internal/1234567890 at 10.10.10.16 G7221/32000 20 ms 640 samples
2011-06-12 10:57:57.617820 [DEBUG] sofia_glue.c:3524 Set 2833 dtmf send/recv
payload to 101
2011-06-12 10:57:57.617820 [DEBUG] sofia.c:4310 (
sofia/internal/1234567890 at 10.10.10.16) State Change CS_NEW -> CS_INIT
2011-06-12 10:57:57.617820 [DEBUG] switch_core_session.c:1021 Send signal
sofia/internal/1234567890 at 10.10.10.16 [BREAK]
2011-06-12 10:57:57.618925 [DEBUG] switch_core_state_machine.c:314 (
sofia/internal/1234567890 at 10.10.10.16) Running State Change CS_INIT
2011-06-12 10:57:57.618925 [DEBUG] switch_core_state_machine.c:338 (
sofia/internal/1234567890 at 10.10.10.16) State INIT
2011-06-12 10:57:57.618925 [DEBUG] mod_sofia.c:83
sofia/internal/1234567890 at 10.10.10.16 SOFIA INIT
2011-06-12 10:57:57.618925 [DEBUG] mod_sofia.c:117 (
sofia/internal/1234567890 at 10.10.10.16) State Change CS_INIT -> CS_ROUTING
2011-06-12 10:57:57.618925 [DEBUG] switch_core_session.c:1021 Send signal
sofia/internal/1234567890 at 10.10.10.16 [BREAK]
2011-06-12 10:57:57.618925 [DEBUG] switch_core_state_machine.c:338 (
sofia/internal/1234567890 at 10.10.10.16) State INIT going to sleep
2011-06-12 10:57:57.618925 [DEBUG] switch_core_state_machine.c:314 (
sofia/internal/1234567890 at 10.10.10.16) Running State Change CS_ROUTING
2011-06-12 10:57:57.618925 [DEBUG] switch_core_state_machine.c:341 (
sofia/internal/1234567890 at 10.10.10.16) State ROUTING
2011-06-12 10:57:57.618925 [DEBUG] mod_sofia.c:140
sofia/internal/1234567890 at 10.10.10.16 SOFIA ROUTING
2011-06-12 10:57:57.618925 [DEBUG] switch_core_state_machine.c:77
sofia/internal/1234567890 at 10.10.10.16 Standard ROUTING
2011-06-12 10:57:57.618925 [INFO] mod_dialplan_xml.c:418 Processing
FreeSWITCH_Conference->1005 in context public
Dialplan: sofia/internal/1234567890 at 10.10.10.16 parsing [public->unloop]
continue=false
Dialplan: sofia/internal/1234567890 at 10.10.10.16 Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1234567890 at 10.10.10.16 Regex (PASS) [unloop]
${sip_looped_call}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1234567890 at 10.10.10.16 Action
deflect(${destination_number})
2011-06-12 10:57:57.620002 [DEBUG] switch_core_state_machine.c:119 (
sofia/internal/1234567890 at 10.10.10.16) State Change CS_ROUTING -> CS_EXECUTE
2011-06-12 10:57:57.620002 [DEBUG] switch_core_session.c:1021 Send signal
sofia/internal/1234567890 at 10.10.10.16 [BREAK]
2011-06-12 10:57:57.620002 [DEBUG] switch_core_state_machine.c:341 (
sofia/internal/1234567890 at 10.10.10.16) State ROUTING going to sleep
2011-06-12 10:57:57.620002 [DEBUG] switch_core_state_machine.c:314 (
sofia/internal/1234567890 at 10.10.10.16) Running State Change CS_EXECUTE
2011-06-12 10:57:57.620002 [DEBUG] switch_core_state_machine.c:348 (
sofia/internal/1234567890 at 10.10.10.16) State EXECUTE
2011-06-12 10:57:57.620002 [DEBUG] mod_sofia.c:226
sofia/internal/1234567890 at 10.10.10.16 SOFIA EXECUTE
2011-06-12 10:57:57.620002 [DEBUG] switch_core_state_machine.c:157
sofia/internal/1234567890 at 10.10.10.16 Standard EXECUTE
EXECUTE sofia/internal/1234567890 at 10.10.10.16 deflect(1005)
2011-06-12 10:57:57.621014 [DEBUG] sofia.c:5004 Process REFER to [
1005 at 10.10.10.16]
*2011-06-12 10:57:57.621014 [ERR] sofia.c:5366 Cannot Blind Transfer 1
Legged calls*
2011-06-12 10:57:57.622041 [DEBUG] sofia.c:4153 Channel
sofia/internal/1005 at 10.10.10.16 entering state [terminated][480]
2011-06-12 10:57:57.622041 [NOTICE] sofia.c:4789 Hangup
sofia/internal/1005 at 10.10.10.16 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE]
2011-06-12 10:57:57.622041 [DEBUG] switch_channel.c:2102 Send signal
sofia/internal/1005 at 10.10.10.16 [KILL]
2011-06-12 10:57:57.622041 [DEBUG] switch_core_session.c:1021 Send signal
sofia/internal/1005 at 10.10.10.16 [BREAK]
2011-06-12 10:57:57.622041 [DEBUG] switch_core_state_machine.c:314 (
sofia/internal/1005 at 10.10.10.16) Running State Change CS_HANGUP
2011-06-12 10:57:57.623126 [DEBUG] switch_core_state_machine.c:499 (
sofia/internal/1005 at 10.10.10.16) State HANGUP
2011-06-12 10:57:57.623126 [DEBUG] mod_sofia.c:408
sofia/internal/1005 at 10.10.10.16 Overriding SIP cause 408 with 480 from the
other leg
2011-06-12 10:57:57.623126 [DEBUG] mod_sofia.c:414 Channel
sofia/internal/1005 at 10.10.10.16 hanging up, cause: NO_USER_RESPONSE
2011-06-12 10:57:57.623126 [DEBUG] switch_ivr_originate.c:3228 Originate
Resulted in Error Cause: 18 [NO_USER_RESPONSE]
2011-06-12 10:57:57.636515 [DEBUG] switch_core_state_machine.c:46
sofia/internal/1005 at 10.10.10.16 Standard HANGUP, cause: NO_USER_RESPONSE
2011-06-12 10:57:57.636515 [DEBUG] switch_core_state_machine.c:499 (
sofia/internal/1005 at 10.10.10.16) State HANGUP going to sleep
2011-06-12 10:57:57.636515 [DEBUG] switch_core_state_machine.c:333 (
sofia/internal/1005 at 10.10.10.16) State Change CS_HANGUP -> CS_REPORTING
2011-06-12 10:57:57.636515 [DEBUG] switch_core_session.c:1021 Send signal
sofia/internal/1005 at 10.10.10.16 [BREAK]
2011-06-12 10:57:57.637437 [DEBUG] switch_core_state_machine.c:314 (
sofia/internal/1005 at 10.10.10.16) Running State Change CS_REPORTING
2011-06-12 10:57:57.637437 [DEBUG] switch_core_state_machine.c:590 (
sofia/internal/1005 at 10.10.10.16) State REPORTING
2011-06-12 10:57:57.637437 [DEBUG] switch_core_state_machine.c:53
sofia/internal/1005 at 10.10.10.16 Standard REPORTING, cause: NO_USER_RESPONSE
2011-06-12 10:57:57.637437 [DEBUG] switch_core_state_machine.c:590 (
sofia/internal/1005 at 10.10.10.16) State REPORTING going to sleep
2011-06-12 10:57:57.637437 [DEBUG] switch_core_state_machine.c:327 (
sofia/internal/1005 at 10.10.10.16) State Change CS_REPORTING -> CS_DESTROY
2011-06-12 10:57:57.637437 [DEBUG] switch_core_session.c:1021 Send signal
sofia/internal/1005 at 10.10.10.16 [BREAK]
2011-06-12 10:57:57.637437 [DEBUG] switch_core_session.c:1164 Session 24 (
sofia/internal/1005 at 10.10.10.16) Locked, Waiting on external entities
2011-06-12 10:57:57.637437 [NOTICE] switch_core_session.c:1182 Session 24 (
sofia/internal/1005 at 10.10.10.16) Ended
2011-06-12 10:57:57.637437 [NOTICE] switch_core_session.c:1184 Close Channel
sofia/internal/1005 at 10.10.10.16 [CS_DESTROY]
2011-06-12 10:57:57.637437 [DEBUG] switch_core_state_machine.c:428 (
sofia/internal/1005 at 10.10.10.16) Running State Change CS_DESTROY
2011-06-12 10:57:57.637437 [DEBUG] switch_core_state_machine.c:439 (
sofia/internal/1005 at 10.10.10.16) State DESTROY
2011-06-12 10:57:57.637437 [DEBUG] mod_sofia.c:341
sofia/internal/1005 at 10.10.10.16 SOFIA DESTROY
2011-06-12 10:57:57.637437 [DEBUG] switch_core_state_machine.c:60
sofia/internal/1005 at 10.10.10.16 Standard DESTROY
2011-06-12 10:57:57.637437 [DEBUG] switch_core_state_machine.c:439 (
sofia/internal/1005 at 10.10.10.16) State DESTROY going to sleep
2011-06-12 10:57:57.720295 [DEBUG] switch_core_session.c:641 Send signal
sofia/internal/1234567890 at 10.10.10.16 [BREAK]
2011-06-12 10:57:57.721283 [NOTICE] switch_core_state_machine.c:185
sofia/internal/1234567890 at 10.10.10.16 has executed the last dialplan
instruction, hanging up.
2011-06-12 10:57:57.721283 [NOTICE] switch_core_state_machine.c:187 Hangup
sofia/internal/1234567890 at 10.10.10.16 [CS_EXECUTE] [NORMAL_CLEARING]
2011-06-12 10:57:57.721283 [DEBUG] switch_channel.c:2102 Send signal
sofia/internal/1234567890 at 10.10.10.16 [KILL]
2011-06-12 10:57:57.722210 [DEBUG] switch_core_session.c:1021 Send signal
sofia/internal/1234567890 at 10.10.10.16 [BREAK]
2011-06-12 10:57:57.722210 [DEBUG] switch_core_state_machine.c:348 (
sofia/internal/1234567890 at 10.10.10.16) State EXECUTE going to sleep
2011-06-12 10:57:57.722210 [DEBUG] switch_core_state_machine.c:314 (
sofia/internal/1234567890 at 10.10.10.16) Running State Change CS_HANGUP
2011-06-12 10:57:57.722210 [DEBUG] switch_core_state_machine.c:499 (
sofia/internal/1234567890 at 10.10.10.16) State HANGUP
2011-06-12 10:57:57.722210 [DEBUG] mod_sofia.c:414 Channel
sofia/internal/1234567890 at 10.10.10.16 hanging up, cause: NORMAL_CLEARING
2011-06-12 10:57:57.722210 [DEBUG] mod_sofia.c:476 Responding to INVITE
with: 480
2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:46
sofia/internal/1234567890 at 10.10.10.16 Standard HANGUP, cause:
NORMAL_CLEARING
2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:499 (
sofia/internal/1234567890 at 10.10.10.16) State HANGUP going to sleep
2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:333 (
sofia/internal/1234567890 at 10.10.10.16) State Change CS_HANGUP ->
CS_REPORTING
2011-06-12 10:57:57.723321 [DEBUG] switch_core_session.c:1021 Send signal
sofia/internal/1234567890 at 10.10.10.16 [BREAK]
2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:314 (
sofia/internal/1234567890 at 10.10.10.16) Running State Change CS_REPORTING
2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:590 (
sofia/internal/1234567890 at 10.10.10.16) State REPORTING
2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:53
sofia/internal/1234567890 at 10.10.10.16 Standard REPORTING, cause:
NORMAL_CLEARING
2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:590 (
sofia/internal/1234567890 at 10.10.10.16) State REPORTING going to sleep
2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:327 (
sofia/internal/1234567890 at 10.10.10.16) State Change CS_REPORTING ->
CS_DESTROY
2011-06-12 10:57:57.723321 [DEBUG] switch_core_session.c:1021 Send signal
sofia/internal/1234567890 at 10.10.10.16 [BREAK]
2011-06-12 10:57:57.723321 [DEBUG] switch_core_session.c:1164 Session 25 (
sofia/internal/1234567890 at 10.10.10.16) Locked, Waiting on external entities
2011-06-12 10:57:57.723321 [NOTICE] switch_core_session.c:1182 Session 25 (
sofia/internal/1234567890 at 10.10.10.16) Ended
2011-06-12 10:57:57.723321 [NOTICE] switch_core_session.c:1184 Close Channel
sofia/internal/1234567890 at 10.10.10.16 [CS_DESTROY]
2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:428 (
sofia/internal/1234567890 at 10.10.10.16) Running State Change CS_DESTROY
2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:439 (
sofia/internal/1234567890 at 10.10.10.16) State DESTROY
2011-06-12 10:57:57.723321 [DEBUG] mod_sofia.c:341
sofia/internal/1234567890 at 10.10.10.16 SOFIA DESTROY
2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:60
sofia/internal/1234567890 at 10.10.10.16 Standard DESTROY
2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:439 (
sofia/internal/1234567890 at 10.10.10.16) State DESTROY going to sleep
##################################################################################
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From michal.bielicki at seventhsignal.de  Sun Jun 12 21:30:35 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Sun, 12 Jun 2011 19:30:35 +0200
Subject: [Freeswitch-users] Invitation to connect on LinkedIn
In-Reply-To: <1517215760.4106057.1307873588760.JavaMail.app@ela4-bed78.prod>
References: <1517215760.4106057.1307873588760.JavaMail.app@ela4-bed78.prod>
Message-ID: <92787CF9-ED2A-4CBE-9431-C5A9CB593B07@seventhsignal.de>

Would you now ....

Am 12.06.2011 um 12:13 schrieb Woody Dickson:

> LinkedIn
> FreeSWITCH,
> 
> I'd like to add you to my professional network on LinkedIn.
> 
> - Woody
> 
> Woody Dickson
> Managing Director at Power Telecom 
> San Francisco Bay Area
> Confirm that you know Woody
> 
> ? 2011, LinkedIn Corporation
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Gesch?ftsf?hrer / CEO

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

Amtsgericht Charlottenburg HRA 44413 B
Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de

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From david.ponzone at ipeva.fr  Sun Jun 12 22:22:27 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sun, 12 Jun 2011 20:22:27 +0200
Subject: [Freeswitch-users] Codec/CDR Error
In-Reply-To: <1307890866.2056.31.camel@gustavo-laptop>
References: <1307890866.2056.31.camel@gustavo-laptop>
Message-ID: <8BDCFAF1-7BF5-4691-AECB-A22DE5E813B4@ipeva.fr>

which FS version do you use ?

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

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

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




Le 12/06/2011 ? 17:01, Gustavo Espeche a ?crit :

> Hi we found an error in freeswitch
> 
> 2011-06-12 10:45:35.662591 [ERR] sofia.c:5323 Codec Error! v=0
> o=Argentina-2 1307890456 1307890458 IN IP4 200.49.30.68
> s=sip call
> c=IN IP4 200.49.30.69
> t=0 0
> m=audio 42410 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> --------------------------------------------
> 2011-06-12 10:48:28.322557 [ERR] sofia.c:5323 Codec Error! v=0
> o=Sonus_UAC 19969 23951 IN IP4 38.105.229.116
> s=SIP Media Capabilities
> c=IN IP4 38.105.229.114
> t=0 0
> m=audio 14578 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> --------------------------------------------
> 2011-06-12 10:50:14.686620 [ERR] sofia.c:5323 Codec Error! v=0
> o=ATSI-B2 188 2 IN IP4 208.37.200.235
> s=sip call
> c=IN IP4 208.37.200.245
> t=0 0
> m=audio 35538 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> --------------------------------------------------------
> 2011-06-12 10:47:52.362623 [ERR] sofia.c:5323 Codec Error! v=0
> o=VoipSwitch 10828 10828 IN IP4 50.30.35.13
> s=VoipSIP
> i=Audio Session
> c=IN IP4 50.30.35.13
> t=0 0
> m=audio 9828 RTP/AVP 18 101
> a=rtpmap:18 G729/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-1
> 
> the problem is that when this error happens FS send the cdr immediately
> but the call still connected, and the provider still with all billing,
> because of the we see a billing difference with our provider.
> The error happens with different provider or customer equipment.
> Some one know if we can configure of freeswitch in some way that the
> call hangup.
> The problem isn't fix the codec error, for us the real problem is that
> the FS do a cdr but the call still connected.
> Thank a lot in advice.
> 
> 
> -- 
> 
> Gustavo Espeche
> EasyIpCall S.R.L.
> www.easyipcall.com
> Bv Mitre 517 24? E
> Cordoba - Argentina
> Te: +54 - 351 - 4280633
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From kheimerl at cs.berkeley.edu  Sun Jun 12 23:55:27 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Sun, 12 Jun 2011 12:55:27 -0700
Subject: [Freeswitch-users] Sofia invite issues
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59DEE58661@cooper>
References: <BANLkTikWTD7bT+kiMdDk1VTOU2XsPU0rcQ@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DEE58301@cooper>
	<BANLkTinvK0rhcqov3SL8aCqp8UL4sJ9Y_Q@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DEF56421@cooper>
	<BANLkTimK5Orq9=12M7qwm8x3qf40tJ-=Aw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DEE58661@cooper>
Message-ID: <BANLkTimW-DMB6H_iDXiRC7nWu6XgnVjs-A@mail.gmail.com>

Just so the robots catch this, Peter was exactly right, and this
resolved the issue.

Thanks!

On Thu, Jun 9, 2011 at 11:54 PM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> In that case, just add the cidr stuff to the global acl "domain" (acl.conf.xml under autoload_configs), this is used by default by the sofia internal profile, or create a new acl the the profile uses.
>
> Something like;
> ? ? ?<node type="allow" cidr="192.168.1.0/24"/>
> ? ? ?<node type="allow" cidr="127.0.0.1/32"/>
>
> This will make all calls from these IP's just to pass in. They won't be detected as any user, so the original From will be used when bridging the outgoing call.
>
> /Peter
>
>
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Kurtis Heimerl
> Skickat: den 10 juni 2011 05:29
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] Sofia invite issues
>
> I think you're right. Thanks! Removing IMSI641104278340883 from that
> domain caused FS to fall back to the previous user, 1301.
>
> Unfortunately, I can't lets the agents auth, or use different IPs, all
> are coming from the same application that doesn't support auth.
>
> I'm going to try adding each to their own unique domain, hopefully
> that'll help. It might not though, considering they're all coming from
> the same IP.
>
> Do you have any other idea how to fix this?
>
> On Thu, Jun 9, 2011 at 1:55 PM, Peter Olsson
> <peter.olsson at visionutveckling.se> wrote:
>> I'm guessing your problem might be related to the fact that you use the cidr attribute on the user's - with the same IP's for all of them. If I understand these correctly (never used it this way myself) they're supposed to be used to authenticate a user for a specific IP, instead of challenge auth, and since the same IP is provided for all of them they might overwrite eachother, and the last one will be the one it matches.
>>
>> Also, read more on http://wiki.freeswitch.org/wiki/Acl#Users.
>>
>> Try removing the cidr attribute, and let the agents auth instead, or use different IP's for the users.
>>
>> /Peter
>> ________________________________________
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Kurtis Heimerl [kheimerl at cs.berkeley.edu]
>> Skickat: den 9 juni 2011 22:21
>> Till: FreeSWITCH Users Help
>> ?mne: Re: [Freeswitch-users] Sofia invite issues
>>
>> Here you go: http://pastebin.freeswitch.org/16468
>>
>> For this packet trace I decided to have profile "C" in the UA
>> directory. This causes FS to send the second invite (which should be
>> A->B) as C->B, rather than B->B, as it does if that UA is not present.
>> ?Neither is correct, this just makes it a little easier to grep.
>>
>> User 1302 first appears in that invite, I don't see any other log
>> items talking about 1302 at all. It's worth noting that 1302's user
>> name (IMSI641104878332498) DOES appear in the SDP packet loaded in the
>> first (A->B) invite, but I'm pretty certain FS should not be using
>> that information for SIP addressing. Again, if I remove user
>> IMSI641104878332498 from the directory, FS just switched to returning
>> with user 1301, it doesn't fix the issue.
>>
>> It's very likely I'm massaging some code that's not commonly used, as
>> I may be making a small mistake here or there in the original SIP
>> invite. These two lines:
>> 2011-06-09 13:11:32.947157 [DEBUG] sofia.c:6674 IP 192.168.1.144
>> Approved by acl "domains[IMSI641104878332498 at 192.168.1.144]". Access
>> Granted.
>> 2011-06-09 13:11:32.947157 [DEBUG] sofia.c:6803 Authenticating user
>> IMSI641104878332498 at 192.168.1.144
>> Are particularly curious. I'm not sure if those are a response to the
>> incoming invite (which would be incorrect) or not. It should be
>> approving user 1300 at 192.168.1.144, if anyone.
>>
>> Thanks!
>>
>> On Thu, Jun 9, 2011 at 12:18 AM, Peter Olsson
>> <peter.olsson at visionutveckling.se> wrote:
>>> Please pastebin a complete debug log, with "sofia global siptrace on", to http://pastebin.freeswitch.org.
>>>
>>> I'm pretty sure this is not a bug in FS, if it was, we would have lots of people complaining...
>>>
>>> /Peter
>>>
>>>
>>> -----Ursprungligt meddelande-----
>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Kurtis Heimerl
>>> Skickat: den 8 juni 2011 23:32
>>> Till: freeswitch-users at lists.freeswitch.org
>>> ?mne: [Freeswitch-users] Sofia invite issues
>>>
>>> Hello Freeswitch-users!
>>>
>>> I'm currently writing a SIP middlebox that intercepts sip messages and
>>> changes their username. The reasons for this are complicated and
>>> somewhat outside of the scope of this discussion. However, I've come
>>> upon a very strange issue: when making a phone-to-phone call across
>>> freeswitch; it is connecting the wrong user. I'm fairly well convinced
>>> this is a bug, but I thought I'd send the issue here and see if it's
>>> anything I'm obviously doing wrong.
>>>
>>> Basically, FS sees a SIP message from a registered UA (call it A)
>>> inviting another registered user to a call (B). This is acked
>>> correctly (a TRYING message). FS then responds by inviting B into a
>>> call with some OTHER user C. C is not mentioned at any point in the
>>> initial SIP messages (as verified by wireshark). Even more strangely,
>>> if I remove user C from FS (by removing their config file) FS responds
>>> to the invite by inviting B into a call with itself. In each case, the
>>> appropriate from header in the invite should be the original caller A.
>>> ?It's worth noting that I am not monkeying with the RTP packets at
>>> all, but my understanding is that SIP signalling shouldn't be affected
>>> by that.
>>>
>>> My general guess is that I'm messing up the naming somehow, and FS is
>>> running an algorithm to guess at who the call originator is. However,
>>> the naming must be roughly correct; Asterisk is able to handle this
>>> call just fine. I've included a sip trace of the second situation (A
>>> calls B, FS invites B to a call with itself)
>>>
>>> For the record A:1300
>>> B:1301
>>> C:IMSI641104878332498
>>>
>>> REGISTER MESSAGE (Which works fine)
>>>
>>> REGISTER sip:192.168.1.144 SIP/2.0
>>> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
>>> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
>>> From: 1300 <sip:1300 at 192.168.1.144>;tag=ftoui
>>> To: 1300 <sip:1300 at 192.168.1.144>
>>> Call-ID: 1032827938 at 192.168.1.144
>>> CSeq: 91 REGISTER
>>> Contact: <sip:1300 at 192.168.1.144:5063>;expires=7200
>>> Max-Forwards: 70
>>> Content-Length: 0
>>>
>>> SIP/2.0 200 OK
>>> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
>>> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
>>> From: 1300 <sip:1300 at 192.168.1.144>;tag=ftoui
>>> To: 1300 <sip:1300 at 192.168.1.144>;tag=jU64NXypQc57F
>>> Call-ID: 1032827938 at 192.168.1.144
>>> CSeq: 91 REGISTER
>>> Contact: <sip:1300 at 192.168.1.144:5063>;expires=7200
>>> Date: Wed, 08 Jun 2011 21:02:04 GMT
>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-71d64e6 2011-06-07 22-30-16 -0700
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>> Supported: timer, precondition, path, replaces
>>> Content-Length: 0
>>>
>>> SIP TRACE (A -> B, B -> B response)
>>>
>>> INITIAL INVITE
>>> INVITE sip:1301 at 192.168.1.144 SIP/2.0
>>> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK77974
>>> From: 1300 <sip:1300 at 192.168.1.144>;tag=bgdqx
>>> To: <sip:1301 at 192.168.1.144>
>>> Call-ID: 1817795092 at 192.168.1.144
>>> CSeq: 485 INVITE
>>> Contact: <sip:1300 at 192.168.1.144:5063>;expires=3600
>>> Content-Type: application/sdp
>>> Max-Forwards: 70
>>> Content-Length: 143
>>>
>>> INVITE ACK
>>> SIP/2.0 100 Trying
>>> Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK77974
>>> From: 1300 <sip:1300 at 192.168.1.144>;tag=bgdqx
>>> To: <sip:1301 at 192.168.1.144>
>>> Call-ID: 1817795092 at 192.168.1.144
>>> CSeq: 485 INVITE
>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-71d64e6 2011-06-07 22-30-16 -0700
>>> Content-Length: 0
>>>
>>> FS INVITE (Note the from address being wrong)
>>> INVITE sip:1301 at 192.168.1.144:5063 SIP/2.0
>>> Via: SIP/2.0/UDP 192.168.1.144;rport;branch=z9hG4bKeeQvcyZ70SDUg
>>> Max-Forwards: 69
>>> From: "1301" <sip:1301 at 192.168.1.144>;tag=rHyS0Z3B61arN
>>> To: <sip:1301 at 192.168.1.144:5063>
>>> Call-ID: 84df47f9-0cb7-122f-13b5-5cff350d9de5
>>> CSeq: 13447852 INVITE
>>> Contact: <sip:mod_sofia at 192.168.1.144:5060>
>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-71d64e6 2011-06-07 22-30-16 -0700
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>> Supported: timer, precondition, path, replaces
>>> Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>>> sla, include-session-description, presence.winfo, message-summary,
>>> refer
>>> Content-Type: application/sdp
>>> Content-Disposition: session
>>> Content-Length: 315
>>> X-FS-Support: update_display
>>> Remote-Party-ID: "1301"
>>> <sip:1301 at 192.168.1.144>;party=calling;screen=yes;privacy=off
>>>
>>> USER CONFIGURATIONS
>>>
>>> 1300.xml
>>> <include>
>>> ?<user id="1300" cidr="192.168.1.0/24,127.0.0.1/32">
>>> ? ?<params>
>>> ? ? ?<param name="accept-blind-reg" value="true"/>
>>> ? ? ?<param name="accept-blind-auth" value="true"/>
>>> ? ? ?<param name="vm-password" value="1000"/>
>>> ? ?</params>
>>> ? ?<variables>
>>> ? ? ?<variable name="toll_allow" value="domestic,international,local"/>
>>> ? ? ?<variable name="accountcode" value="1300"/>
>>> ? ? ?<variable name="user_context" value="default"/>
>>> ? ? ?<variable name="effective_caller_id_name" value="1300"/>
>>> ? ? ?<variable name="effective_caller_id_number" value="1300"/>
>>> ? ? ?<variable name="outbound_caller_id_name" value="1300"/>
>>> ? ? ?<variable name="outbound_caller_id_number" value="1300"/>
>>> ? ? ?<variable name="callgroup" value="techsupport"/>
>>> ? ?</variables>
>>> ?</user>
>>> </include>
>>>
>>> 1301.xml
>>> <include>
>>> ?<user id="1301" cidr="192.168.1.0/24,127.0.0.1/32">
>>> ? ?<params>
>>> ? ? ?<param name="accept-blind-reg" value="true"/>
>>> ? ? ?<param name="accept-blind-auth" value="true"/>
>>> ? ? ?<param name="vm-password" value="1000"/>
>>> ? ?</params>
>>> ? ?<variables>
>>> ? ? ?<variable name="toll_allow" value="domestic,international,local"/>
>>> ? ? ?<variable name="accountcode" value="1301"/>
>>> ? ? ?<variable name="user_context" value="default"/>
>>> ? ? ?<variable name="effective_caller_id_name" value="1301"/>
>>> ? ? ?<variable name="effective_caller_id_number" value="1301"/>
>>> ? ? ?<variable name="outbound_caller_id_name" value="1301"/>
>>> ? ? ?<variable name="outbound_caller_id_number" value="1301"/>
>>> ? ? ?<variable name="callgroup" value="techsupport"/>
>>> ? ?</variables>
>>> ?</user>
>>> </include>
>>>
>>> IMSI641104878332498.xml
>>> <include>
>>> ?<user id="IMSI641104878332498" cidr="192.168.1.0/24,127.0.0.1/32"
>>> number-alias="1302">
>>> ? ?<params>
>>> ? ? ?<param name="accept-blind-reg" value="true"/>
>>> ? ? ?<param name="accept-blind-auth" value="true"/>
>>> ? ? ?<param name="vm-password" value="1000"/>
>>> ? ?</params>
>>> ? ?<variables>
>>> ? ? ?<variable name="toll_allow" value="domestic,international,local"/>
>>> ? ? ?<variable name="accountcode" value="1302"/>
>>> ? ? ?<variable name="user_context" value="default"/>
>>> ? ? ?<variable name="effective_caller_id_name" value="IMSI641104878332498"/>
>>> ? ? ?<variable name="effective_caller_id_number" value="1302"/>
>>> ? ? ?<variable name="outbound_caller_id_name" value="IMSI641104878332498"/>
>>> ? ? ?<variable name="outbound_caller_id_number" value="1302"/>
>>> ? ? ?<variable name="callgroup" value="techsupport"/>
>>> ? ?</variables>
>>> ?</user>
>>> </include>
>>>
>>> DIALPLAN
>>>
>>> ? ?<extension name="Local Numbers">
>>> ? ? ?<condition field="destination_number" expression="^130[0-9]$">
>>> ? ? ? ?<!--<action application="bridge"
>>> data="user/${user_data(${destination_number}@${domain_name} attr
>>> id)}"/> -->
>>> ? ? ? ?<action application="bridge" data="user/${destination_number}"/>
>>> ? ? ?</condition>
>>> ? ?</extension>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> !DSPAM:4df1916532761203114732!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From freeswitch at aastral.net  Mon Jun 13 00:48:19 2011
From: freeswitch at aastral.net (Bill W.)
Date: Sun, 12 Jun 2011 16:48:19 -0400
Subject: [Freeswitch-users] start_dtmf_generate during early media?
Message-ID: <1QVra4-0008Vr-8x@mail.aastral.net>

Hey everyone,

I'm trying to terminate a call over a voip carrier to an AT&T conference
bridge.

AT&T (who, of all people, should know better) are playing the prompts
for the conference bridge over early media.  And of course, you can't
connect to the conf. bridge because no DTMF is transferred (because the
call isn't answered yet).

So the question is, is there a way to force FreeSWITCH to transcode 2833
DTMF from the A-leg and send inband DTMF on the B-leg while the B-leg is
in early-media state?

I see there is the start_dtmf_generate application, but will this work
if the b-leg is still in the early-media state?

Thanks!


From david.ponzone at ipeva.fr  Mon Jun 13 03:52:36 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 13 Jun 2011 01:52:36 +0200
Subject: [Freeswitch-users] g729 licenses are not valid anymore after
	upgrade
In-Reply-To: <37601782DCD4427186EF316CA19FD65A@omni1.local>
References: <9335B0A6-5190-4B14-86FE-EE515AF0B97A@ipeva.fr>
	<37601782DCD4427186EF316CA19FD65A@omni1.local>
Message-ID: <DB8B72C4-EFF4-41C5-9CFB-901A0EA2713A@ipeva.fr>

Anestis,

absolutely.
I tried that yesterday but it failed.
I supposed I screwed up somewhere because I just did it again, and it's ok now.

Thanks

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

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

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




Le 12/06/2011 ? 14:18, Anestis Mavro a ?crit :

> Just activate the license again. I?ve had the same problem a few weeks ago?
>  
> Regards
> Anestis
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Ponzone
> Sent: Sunday, June 12, 2011 2:35 AM
> To: FreeSWITCH Users Help
> Subject: [Freeswitch-users] g729 licenses are not valid anymore after upgrade
>  
> I just upgraded from a 6 months GIT version to current, and the latest mod_com_g729 does not seem to accept my G729 licenses anymore.
>  
> The error message is:
> Unrecognized resource G729A/0
>  
> If I revert back to the previous version (with mod_com_g729 158), it works fine.
>  
> Is there something that was changed in the licenses ?
>  
> Thanks
>  
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>  
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>  
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>  
> 
> 
>  
> 
> 
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 5054 (20100423) __________
> 
> The message was checked by ESET NOD32 Antivirus.
> 
> http://www.eset.com
> 
> 
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 5054 (20100423) __________
> 
> The message was checked by ESET NOD32 Antivirus.
> 
> http://www.eset.com
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at ipeva.fr  Mon Jun 13 03:54:45 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 13 Jun 2011 01:54:45 +0200
Subject: [Freeswitch-users] session:read() not timeouting
In-Reply-To: <9911DF4D-FC78-4F88-AB15-2FAB939F3A59@ipeva.fr>
References: <9911DF4D-FC78-4F88-AB15-2FAB939F3A59@ipeva.fr>
Message-ID: <37948572-D5DC-446D-B6E5-B2AC48523B09@ipeva.fr>

I upgraded to latest GIT, so I can now answer to myself: it's fixed.
Something was wrong in Dec 1st 2010 version with application read, and is fixed now.

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

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

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




Le 11/06/2011 ? 14:06, David Ponzone a ?crit :

> All,
> 
> When using session:read() in a LUA script called from dialplan, the interdigit timeout works fine on my test box (with GIT from end of May, and a 1000hz timer).
> On my prod box, the timeout does not trigger the end of the read app. The terminating character is the only way to end the read() normally.
> The prod box is using a GIT from December 2010, and a bad timer (250Hz).
> 
> Should I suppose the issue comes from the timer or from the old version on the prod box (I am upgrading ASAP) ?
> 
> Thanks
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 

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From qingquan at globalroam.com  Mon Jun 13 06:31:24 2011
From: qingquan at globalroam.com (qingquan luo)
Date: Mon, 13 Jun 2011 10:31:24 +0800
Subject: [Freeswitch-users] in bypass_media mode,
	2833 payload type modified by freeswitch!
Message-ID: <BANLkTikVN7ORVyF79rX7moeDuJZafYfyVQ@mail.gmail.com>

Hi All,

  I use freeswitch to bridge one incoming call to other target phone number.
  I use bypass_media mode.
  So the rtp is not go through the freeswitch,
  By I notice that When LegB reply 183 or 200 SDP message. Freeswitch
change it 2833 telephone-event payload type. and forwarding the message to
caller. This make the caller use wrong dtmf payload type to send dtmf

  What wrong with that?  How can fix it?
  Any information or help is welcome.


Thanks



Best Regards


Qingquan
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From michael.raistrick at googlemail.com  Mon Jun 13 10:55:10 2011
From: michael.raistrick at googlemail.com (Mike Raistrick)
Date: Mon, 13 Jun 2011 07:55:10 +0100
Subject: [Freeswitch-users] UK ISDN outgoing CLI (was Re: FreeTDM -
 Sangoma B700 - ISDN connection questions - UK)
In-Reply-To: <4DF1F07A.5000200@earthspike.net>
References: <4DED4854.6010203@earthspike.net>
	<BANLkTimP23FVs8Vcg+pbxnfBucrQoJur2w@mail.gmail.com>
	<4DF10C83.6020803@earthspike.net>
	<BANLkTikCEKqPTFZLDxF+h940RnmizHYLBA@mail.gmail.com>
	<4DF1F07A.5000200@earthspike.net>
Message-ID: <BANLkTin5+puYLD7OtreiRxQZ5i8xd5KppQ@mail.gmail.com>

I've no direct experience with BT BRA ISDN (plenty with other providers) -
but I don't think that CLIP / COLP will be the issue.

CLIP is an inbound service - and determines whether your equipment is
presented with the number of the person calling you (and if they allow it).

Although COLP is an outbound service, it is used to display the number of
the line that you are connected through to (which might not be the same as
the number you dialled - e.g. 0800 numbers).

Whilst we're on supplementary services, CLIR is also an outbound service and
allows you to specify whether the CLI that you send (or the one that BT
uses) can be shown to the person you call.

Some terminogy: ISDN Basic Rate differentiates Direct Dial In (DDI's) and
Multiple Subscriber Numbers (MSNs) for incoming services.

DDIs are designed for PBXs and are associated with a range of numbers in a
block. The service provider sends you a called party number that can range
either from the last 'X' digits through to the whole number - this is
normally specified when the service is ordered and then all calls are
delivered the same way.

With DDIs - when you send an outgoing call you should be able to send the
extension or the whole number - in either case you should mark the Type Of
Number field accordingly (and if in national format - not include the '0'
for the number itself). In case that you send the extension digits - the
service provider adds this to the DDI 'stem' to get the full number. With
the full number (either derived from the extension, or as you sent it) - the
provider should then check that the number is in the allocated DDI range -
if it is it will allow it through, if not it will usually send the root DDI
number.

MSNs are unique to BRA - they are completely separate telephone numbers and
were originally designed to allow multiple BRA phones to be off the same
ISDN bus and have their own phone numbers. This means that the full number
will always be sent to you, and be expected from you for called and calling
numbers (still with the right Type Of Number set..).

Summary - with either DDI or MSNs - you should be OK to send the whole
number to BT as the calling party - minus the '0' and with TON set to
'national'.

Finally - if you do have the CLIR supplementary service, make sure that you
set the presentation indicator to 'allowed' - default should normally be to
assume allowed unless set differently - but to be on the safe side.

Mike



On Fri, Jun 10, 2011 at 11:22 AM, John <freeswitch at earthspike.net> wrote:

> Shouldbe,
>
> You won't be burned at the stake for heresy and are probably right that
> this is the problem, as we have had delays in getting CLIP implemented.
>  From your comment I assume that COLP is not automatically enabled (even
> though I am paying for additional SNDDIs) but has to be ordered
> specifically.  Our service provider is on to it, and I am waiting till
> CLIP/COLP is applied before doing any more testing.  I'll post here with
> the results.
>
> John
>
> On 10/06/11 10:08, shouldbe q931 wrote:
> > On Thu, Jun 9, 2011 at 7:10 PM, John<freeswitch at earthspike.net>  wrote:
> >> [I've forked this thread as it really should have been 3 separate
> >> threads to start with.]
> >>
> >> Thanks,
> >>
> >> I have tried every combination I can think of, except 10 digits which I
> >> will give a go.  That makes sense as in the general case it would permit
> >> non-geographical numbers to be presented.
> >>
> >> The lines are provided as 2 x ISDN2e with a lead number covering all 4
> >> channels.  The NTE8s are labelled 1-2 and 3-4, so not separate
> >> individual lines, I thnk.
> >>
> >> We currently have a Single Number DDI in addition, and hope to increase
> >> that to 2 very soon; the 3 numbers between them will serve 3 separate
> >> business units, so we want to be able to present the correct outbound
> >> CLI from each business unit.  We also have a remote call forward from
> >> another number to the lead number, but presenting that really would be
> >> CLI spoofing, so I am not expecting that to be permissible. But I would
> >> expect those that are directly allocated to the 2 lines/4 channels to be
> >> allowed to be presented.
> >>
> >> John
> >>
> >> On 09/06/11 08:27, shouldbe q931 wrote:
> >>> On Mon, Jun 6, 2011 at 10:36 PM, John<freeswitch at earthspike.net>
>  wrote:
> >>>> Hello,
> >>>>
> >>> <snipped for brevity>
> >>>> 2. We have 2 Single Number DDI numbers configured over the 4 channels.
> >>>> I want to set up outgoing calls so that they can appear to come from
> >>>> either of the two numbers, but at present the outgoing CLI appears to
> be
> >>>> overridden by the telco (BT) to only use one of the numbers.  Has
> anyone
> >>>> got this working in UK, and what is the format for the
> >>>> outbound_caller_id_number: last 6 digits or full 11 digits?  I note
> that
> >>>> the inbound called number is only the last 6 digits.
> >>>>
> >>> <snipped for brevity>
> >>>> Thanks for all the great support I have already got just from editing
> my
> >>>> Wiki User page; this is a friendly group!
> >>>>
> >>>> John
> >>>>
> >>> I can only comment on the BT part.
> >>>
> >>> If they are two individual lines, you're out of luck with BT, as they
> >>> do not allow CLI spoofing, which this would be.
> >>>
> >>> I would suggest sending BT the same number of digits that they send
> >>> you, if that doesn't work try 10 digits. If  you are not sending the
> >>> "correct" number of digits for a valid DDI on the trunk, then they
> >>> will send the lead number for the trunk.
> >>>
> >>> Cheers
> >>>
> >>> _______________________________________________
> > I know that this might sound like heresy :-)  but have you checked
> > spoken to BT and asked them if CLIP and CLOP on the lines ?
> >
> > I think CLOP is outbound and CLIP is inbound, but I might have it the
> > wrong way round...
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gustavo.espeche at easyipcall.com  Mon Jun 13 14:54:51 2011
From: gustavo.espeche at easyipcall.com (Gustavo Espeche)
Date: Mon, 13 Jun 2011 07:54:51 -0300
Subject: [Freeswitch-users] Codec/CDR Error
In-Reply-To: <mailman.10432.1307908600.24555.freeswitch-users@lists.freeswitch.org>
References: <mailman.10432.1307908600.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <1307962491.2216.4.camel@gustavo-laptop>

the freeswitch version is FreeSWITCH Version 1.0.head (git-c4a2faf
2011-06-10 14-36-35 -0400), and in this version we can put the FS in
debug too.

-- 
Gustavo Espeche
EasyIpCall S.R.L.
www.easyipcall.com
Bv Mitre 517 24? E
Cordoba - Argentina
Te: +54 - 351 - 4280633


which FS version do you use ? 
David Ponzone  Direction Technique email: david.ponzone at ipeva.fr tel:
   01 74 03 18 97 gsm:   06 66 98 76 34 

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

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


Le 12/06/2011 ? 17:01, Gustavo Espeche a ?crit : 
> Hi we found an error in freeswitch
> 
> 2011-06-12 10:45:35.662591 [ERR] sofia.c:5323 Codec Error! v=0
> o=Argentina-2 1307890456 1307890458 IN IP4 200.49.30.68
> s=sip call
> c=IN IP4 200.49.30.69
> t=0 0
> m=audio 42410 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> --------------------------------------------
> 2011-06-12 10:48:28.322557 [ERR] sofia.c:5323 Codec Error! v=0
> o=Sonus_UAC 19969 23951 IN IP4 38.105.229.116
> s=SIP Media Capabilities
> c=IN IP4 38.105.229.114
> t=0 0
> m=audio 14578 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> --------------------------------------------
> 2011-06-12 10:50:14.686620 [ERR] sofia.c:5323 Codec Error! v=0
> o=ATSI-B2 188 2 IN IP4 208.37.200.235
> s=sip call
> c=IN IP4 208.37.200.245
> t=0 0
> m=audio 35538 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> --------------------------------------------------------
> 2011-06-12 10:47:52.362623 [ERR] sofia.c:5323 Codec Error! v=0
> o=VoipSwitch 10828 10828 IN IP4 50.30.35.13
> s=VoipSIP
> i=Audio Session
> c=IN IP4 50.30.35.13
> t=0 0
> m=audio 9828 RTP/AVP 18 101
> a=rtpmap:18 G729/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-1
> 
> the problem is that when this error happens FS send the cdr
> immediately
> but the call still connected, and the provider still with all billing,
> because of the we see a billing difference with our provider.
> The error happens with different provider or customer equipment.
> Some one know if we can configure of freeswitch in some way that the
> call hangup.
> The problem isn't fix the codec error, for us the real problem is that
> the FS do a cdr but the call still connected.
> Thank a lot in advice.
> 
> 
> -- 
> 
> Gustavo Espeche
> EasyIpCall S.R.L.
> www.easyipcall.com
> Bv Mitre 517 24? E
> Cordoba - Argentina
> Te: +54 - 351 - 4280633
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From gustavo.espeche at easyipcall.com  Mon Jun 13 14:59:34 2011
From: gustavo.espeche at easyipcall.com (Gustavo Espeche)
Date: Mon, 13 Jun 2011 07:59:34 -0300
Subject: [Freeswitch-users] Sangoma Card D100
Message-ID: <1307962774.2216.6.camel@gustavo-laptop>

Hi Some one can do this card work, we was buy it 2 week ago and nor
Sangoma support can't it work, the card don't do transcoding, if someone
can do it work please let us know.

-- 

Gustavo Espeche
EasyIpCall S.R.L.
www.easyipcall.com
Bv Mitre 517 24? E
Cordoba - Argentina
Te: +54 - 351 - 4280633



From david.ponzone at ipeva.fr  Mon Jun 13 15:30:15 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 13 Jun 2011 13:30:15 +0200
Subject: [Freeswitch-users] Sangoma Card D100
In-Reply-To: <1307962774.2216.6.camel@gustavo-laptop>
References: <1307962774.2216.6.camel@gustavo-laptop>
Message-ID: <F5F9D15F-6FC0-45C1-96FE-7965D0A3A170@ipeva.fr>

May you explain exactly what makes you think it does not work and what are the steps you followed to install it ?

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

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

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




Le 13/06/2011 ? 12:59, Gustavo Espeche a ?crit :

> Hi Some one can do this card work, we was buy it 2 week ago and nor
> Sangoma support can't it work, the card don't do transcoding, if someone
> can do it work please let us know.
> 
> -- 
> 
> Gustavo Espeche
> EasyIpCall S.R.L.
> www.easyipcall.com
> Bv Mitre 517 24? E
> Cordoba - Argentina
> Te: +54 - 351 - 4280633
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From andy at fabulous4.co.uk  Mon Jun 13 17:48:02 2011
From: andy at fabulous4.co.uk (Andy Ayers)
Date: Mon, 13 Jun 2011 14:48:02 +0100
Subject: [Freeswitch-users] SIP Option Method
Message-ID: <00cc01cc29d0$855d4380$9017ca80$@fabulous4.co.uk>

Hi folks,

 

I've been asked by my telecoms provider if my switch supports the SIP OPTION
METHOD. I'm running freeswitch 1.0.6 on a debian linux server. Can you
confirm if this is supported or not?

 

Many thanks

Andy

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From rhuddleston at gmail.com  Mon Jun 13 17:51:49 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Mon, 13 Jun 2011 09:51:49 -0400
Subject: [Freeswitch-users] SIP Option Method
In-Reply-To: <00cc01cc29d0$855d4380$9017ca80$@fabulous4.co.uk>
References: <00cc01cc29d0$855d4380$9017ca80$@fabulous4.co.uk>
Message-ID: <1c4b01cc29d1$0b848680$228d9380$@com>

RTFM / JFGI

 

http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Andy
Ayers
Sent: Monday, June 13, 2011 9:48 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] SIP Option Method

 

Hi folks,

 

I've been asked by my telecoms provider if my switch supports the SIP OPTION
METHOD. I'm running freeswitch 1.0.6 on a debian linux server. Can you
confirm if this is supported or not?

 

Many thanks

Andy

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From steveayre at gmail.com  Mon Jun 13 18:24:14 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 13 Jun 2011 15:24:14 +0100
Subject: [Freeswitch-users] SIP Option Method
In-Reply-To: <00cc01cc29d0$855d4380$9017ca80$@fabulous4.co.uk>
References: <00cc01cc29d0$855d4380$9017ca80$@fabulous4.co.uk>
Message-ID: <BANLkTinbnniWyYhczmNSPv6Vy+3Kn1R9Eg@mail.gmail.com>

Yes


On 13 June 2011 14:48, Andy Ayers <andy at fabulous4.co.uk> wrote:

> Hi folks,
>
>
>
> I?ve been asked by my telecoms provider if my switch supports the SIP
> OPTION METHOD. I?m running freeswitch 1.0.6 on a debian linux server. Can
> you confirm if this is supported or not?
>
>
>
> Many thanks
>
> Andy
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From wstephen80 at gmail.com  Mon Jun 13 18:28:51 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Mon, 13 Jun 2011 16:28:51 +0200
Subject: [Freeswitch-users] Sangoma Card D100
In-Reply-To: <1307962774.2216.6.camel@gustavo-laptop>
References: <1307962774.2216.6.camel@gustavo-laptop>
Message-ID: <BANLkTimyxw6jR8zymAba8fm_h-yUE-R9Tg@mail.gmail.com>

Hi, I'm using a Sangoma D500 transcoding card.
No problem to install and configure the board: is zero config board, you
have only to load the sangoma codec freeswitch module.
As David Ponzone say, you have to do a detailed explanation of your trouble.
Stephen

On Mon, Jun 13, 2011 at 12:59 PM, Gustavo Espeche <
gustavo.espeche at easyipcall.com> wrote:

> Hi Some one can do this card work, we was buy it 2 week ago and nor
> Sangoma support can't it work, the card don't do transcoding, if someone
> can do it work please let us know.
>
> --
>
> Gustavo Espeche
> EasyIpCall S.R.L.
> www.easyipcall.com
> Bv Mitre 517 24? E
> Cordoba - Argentina
> Te: +54 - 351 - 4280633
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kris at kriskinc.com  Mon Jun 13 18:46:36 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Mon, 13 Jun 2011 10:46:36 -0400
Subject: [Freeswitch-users] in bypass_media mode,
 2833 payload type modified by freeswitch!
In-Reply-To: <BANLkTikVN7ORVyF79rX7moeDuJZafYfyVQ@mail.gmail.com>
References: <BANLkTikVN7ORVyF79rX7moeDuJZafYfyVQ@mail.gmail.com>
Message-ID: <BANLkTikJO6JUtJ6KEyqJ1wyFeLQ9sUiawA@mail.gmail.com>

Please provide a full console log with sip trace enabled and upload it
to http://pastebin.freeswitch.org

sofia global siptrace on

On Sun, Jun 12, 2011 at 10:31 PM, qingquan luo <qingquan at globalroam.com> wrote:
> Hi All,
>
>   I use freeswitch to bridge one incoming call to other target phone number.
>   I use bypass_media mode.
>   So the rtp is not go through the freeswitch,
>   By I notice that When LegB reply 183 or 200 SDP message. Freeswitch
> change it 2833 telephone-event payload type. and forwarding the message to
> caller. This make the caller use wrong dtmf payload type to send dtmf
>
>   What wrong with that?  How can fix it?
>   Any information or help is welcome.
>
>
> Thanks
>
> Best Regards
>
>
> Qingquan
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner


From ankitwalia4u at gmail.com  Mon Jun 13 15:08:19 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Mon, 13 Jun 2011 16:38:19 +0530
Subject: [Freeswitch-users] X-Lite giving error while connecting to
	FreeSwitch Server
Message-ID: <BANLkTimf-E48+fCz7tUceRkR1zP+HvSXzw@mail.gmail.com>

Hello all,

I am very new to FreeSwitch and VOIP Technology.

I installed the FreeSwitch and X-Lite on my Window 7 OS.

I am using default FreeSwitch configuration. Nothing changed except adding
Text to Speech module.

When I am trying connecting to FreeSwitch Sever on localhost from X-Lite
Softphone. I am getting the error on X-lite
Account failed to enable.
Account 1001 could not be enabled.
Problem at server(SIP error 503). Try Again Later

I have given the account setting at X-Lite as below:
Account Name: 1001
Protocol: SIP
UserID: 1001
Domain: localhost
Password: 1234
Display Name: Test User
Authorization Name: 1001

Domain Proxy:
Register with Domain and receive calls: Checked
Send Outbound via: DOmain

I dont know what wrong I am doing and How I should debug this issue.

Please help.

Thanks
Ankit
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From msc at freeswitch.org  Mon Jun 13 19:11:08 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 13 Jun 2011 08:11:08 -0700
Subject: [Freeswitch-users] Codec/CDR Error
In-Reply-To: <1307962491.2216.4.camel@gustavo-laptop>
References: <mailman.10432.1307908600.24555.freeswitch-users@lists.freeswitch.org>
	<1307962491.2216.4.camel@gustavo-laptop>
Message-ID: <BANLkTi=Gpf2rhSMf4_WKBSCftcUfG+hEyw@mail.gmail.com>

get a full debug trace w/ sip trace and put it on pastebin.freeswitch.org.
-MC

On Mon, Jun 13, 2011 at 3:54 AM, Gustavo Espeche <
gustavo.espeche at easyipcall.com> wrote:

> the freeswitch version is FreeSWITCH Version 1.0.head (git-c4a2faf
> 2011-06-10 14-36-35 -0400), and in this version we can put the FS in
> debug too.
>
> --
> Gustavo Espeche
> EasyIpCall S.R.L.
> www.easyipcall.com
> Bv Mitre 517 24? E
> Cordoba - Argentina
> Te: +54 - 351 - 4280633
>
>
> which FS version do you use ?
> David Ponzone  Direction Technique email: david.ponzone at ipeva.fr tel:
>   01 74 03 18 97 gsm:   06 66 98 76 34
>
> Service Client IPeva tel:      0811 46 26 26 www.ipeva.fr  -
>  www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion non autoris?e est interdite. Tout message ?lectronique est
> susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de
> ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et
> d'avertir l'exp?diteur.
>
>
> Le 12/06/2011 ? 17:01, Gustavo Espeche a ?crit :
> > Hi we found an error in freeswitch
> >
> > 2011-06-12 10:45:35.662591 [ERR] sofia.c:5323 Codec Error! v=0
> > o=Argentina-2 1307890456 1307890458 IN IP4 200.49.30.68
> > s=sip call
> > c=IN IP4 200.49.30.69
> > t=0 0
> > m=audio 42410 RTP/AVP 18 101
> > a=rtpmap:18 G729/8000
> > a=fmtp:18 annexb=no
> > a=rtpmap:101 telephone-event/8000
> > --------------------------------------------
> > 2011-06-12 10:48:28.322557 [ERR] sofia.c:5323 Codec Error! v=0
> > o=Sonus_UAC 19969 23951 IN IP4 38.105.229.116
> > s=SIP Media Capabilities
> > c=IN IP4 38.105.229.114
> > t=0 0
> > m=audio 14578 RTP/AVP 18 101
> > a=rtpmap:18 G729/8000
> > a=fmtp:18 annexb=no
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-15
> > a=ptime:20
> > --------------------------------------------
> > 2011-06-12 10:50:14.686620 [ERR] sofia.c:5323 Codec Error! v=0
> > o=ATSI-B2 188 2 IN IP4 208.37.200.235
> > s=sip call
> > c=IN IP4 208.37.200.245
> > t=0 0
> > m=audio 35538 RTP/AVP 18 101
> > a=rtpmap:18 G729/8000
> > a=fmtp:18 annexb=no
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-15
> > --------------------------------------------------------
> > 2011-06-12 10:47:52.362623 [ERR] sofia.c:5323 Codec Error! v=0
> > o=VoipSwitch 10828 10828 IN IP4 50.30.35.13
> > s=VoipSIP
> > i=Audio Session
> > c=IN IP4 50.30.35.13
> > t=0 0
> > m=audio 9828 RTP/AVP 18 101
> > a=rtpmap:18 G729/8000/1
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-1
> >
> > the problem is that when this error happens FS send the cdr
> > immediately
> > but the call still connected, and the provider still with all billing,
> > because of the we see a billing difference with our provider.
> > The error happens with different provider or customer equipment.
> > Some one know if we can configure of freeswitch in some way that the
> > call hangup.
> > The problem isn't fix the codec error, for us the real problem is that
> > the FS do a cdr but the call still connected.
> > Thank a lot in advice.
> >
> >
> > --
> >
> > Gustavo Espeche
> > EasyIpCall S.R.L.
> > www.easyipcall.com
> > Bv Mitre 517 24? E
> > Cordoba - Argentina
> > Te: +54 - 351 - 4280633
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Jun 13 19:20:01 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 13 Jun 2011 08:20:01 -0700
Subject: [Freeswitch-users] Listen to the conversation between a user
 and an IVR
In-Reply-To: <33095823FD21DF429B481B5163264B7950AC643016@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950AC643016@VMBX102.ihostexchange.net>
Message-ID: <BANLkTin2nTR0Dpimy5WEN1kEx3T2p5H5mw@mail.gmail.com>

Whom do you wish to have listening to that call? You can use the eavesdrop
function if you wish. You can also use the telecast feature of mod_xml_rpc
to listen to live calls via a browser...

-MC

On Sat, Jun 11, 2011 at 8:22 AM, Yungwei Chen <yungwei at resolvity.com> wrote:

> Hi,
>
> I'm wondering if there's a way to listen to the conversation between a user
> and an IVR (javascript programs).
>
> Here's the background info.
> I have a javascript that first makes an outbound call to someone. And When
> the call is answered, the call is then connected another extension, which
> runs an IVR application written in javascript.
> I want to be able to listen to the live conversation between the person and
> the IVR.
>
> Is this supported? If so, how can I do that in javascript? Thanks.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Jun 13 19:27:25 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 13 Jun 2011 08:27:25 -0700
Subject: [Freeswitch-users] Originating a call from a confeence is not
 working ?
In-Reply-To: <BANLkTi=_ogA9pig3HTQCoG2mWB9P_a_8ng@mail.gmail.com>
References: <BANLkTi=_ogA9pig3HTQCoG2mWB9P_a_8ng@mail.gmail.com>
Message-ID: <BANLkTinC6WadzA41QH7fvXu-w41cv5oh7Q@mail.gmail.com>

Just curious - Try this and let us know if it behaves differently:

conference 3001-10.10.10.16 dial
{originate_timeout=30}<%7Boriginate_timeout=30%7Dsofia/internal/1005 at 135.254.210.166>user/1005
1234567890
FreeSWITCH_Conference

-MC

On Sun, Jun 12, 2011 at 8:32 AM, Narendra Sirugudi <nsirugudi at gmail.com>wrote:

> Hi,
>
> I am trying to use the dial-out feature of the conference application of
> freeswitch.
> I am using freeswtich version 1.0.6.
>
> As i understand the conference gets created when the first user joins.
> Hence i made one user with number 1004 join the conference.
>
> freeswitch at internal> conference list
> Conference 3001-10.10.10.16 (1 member rate: 8000)
> 6;sofia/internal/1004 at 10.10.10.16
> ;08178796-a06e-4ee8-81fb-1f0cc53f8fb5;1004;1004;hear|speak|floor;0;0;300
>
> Now i dail out another user 1005 using the command :
> freeswitch at internal> conference 3001-10.10.10.16 dial
> {originate_timeout=30}sofia/internal/1005 at 10.10.10.16<%7Boriginate_timeout=30%7Dsofia/internal/1005 at 135.254.210.166>1234567890 FreeSWITCH_Conference
>  This does not work. I observe the following errors in the fs_cli logs:
>
> *2011-06-12 10:57:57.621014 [ERR] sofia.c:5366 Cannot Blind Transfer 1
> Legged calls*
> can anyone tell what could be going wrong ?
>
> thanks
> --naren
>
> The complete logs are given below:
>
> #############################################################
> freeswitch at internal> conference 3001-10.10.10.16 dial
> {originate_timeout=30}sofia/internal/1005 at 10.10.10.16 1234567890
> FreeSWITCH_Conference
> Call Requested: result: [NO_USER_RESPONSE]
> 2011-06-12 10:57:57.606002 [DEBUG] switch_ivr_originate.c:1885 variable
> string 0 = [ignore_early_media=true]
> freeswitch at internal> 2011-06-12 10:57:57.606002 [DEBUG]
> switch_ivr_originate.c:1885 variable string 1 = [originate_timeout=30]
> 2011-06-12 10:57:57.606002 [NOTICE] switch_channel.c:669 New Channel
> sofia/internal/1005 at 10.10.10.16 [a77137ad-ad1a-45e9-ac27-88a996cd865d]
> 2011-06-12 10:57:57.606002 [DEBUG] mod_sofia.c:3384 (
> sofia/internal/1005 at 10.10.10.16) State Change CS_NEW -> CS_INIT
> 2011-06-12 10:57:57.606002 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/1005 at 10.10.10.16 [BREAK]
> 2011-06-12 10:57:57.607036 [DEBUG] switch_core_state_machine.c:314 (
> sofia/internal/1005 at 10.10.10.16) Running State Change CS_INIT
> 2011-06-12 10:57:57.607036 [DEBUG] switch_core_state_machine.c:338 (
> sofia/internal/1005 at 10.10.10.16) State INIT
> 2011-06-12 10:57:57.607036 [DEBUG] mod_sofia.c:83
> sofia/internal/1005 at 10.10.10.16 SOFIA INIT
> 2011-06-12 10:57:57.607036 [DEBUG] mod_sofia.c:117 (
> sofia/internal/1005 at 10.10.10.16) State Change CS_INIT -> CS_ROUTING
> 2011-06-12 10:57:57.607036 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/1005 at 10.10.10.16 [BREAK]
> 2011-06-12 10:57:57.607036 [DEBUG] switch_core_state_machine.c:338 (
> sofia/internal/1005 at 10.10.10.16) State INIT going to sleep
> 2011-06-12 10:57:57.607036 [DEBUG] switch_core_state_machine.c:314 (
> sofia/internal/1005 at 10.10.10.16) Running State Change CS_ROUTING
> 2011-06-12 10:57:57.607036 [DEBUG] switch_core_state_machine.c:341 (
> sofia/internal/1005 at 10.10.10.16) State ROUTING
> 2011-06-12 10:57:57.607036 [DEBUG] mod_sofia.c:140
> sofia/internal/1005 at 10.10.10.16 SOFIA ROUTING
> 2011-06-12 10:57:57.607036 [DEBUG] switch_ivr_originate.c:66 (
> sofia/internal/1005 at 10.10.10.16) State Change CS_ROUTING ->
> CS_CONSUME_MEDIA
> 2011-06-12 10:57:57.607036 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/1005 at 10.10.10.16 [BREAK]
> 2011-06-12 10:57:57.607036 [DEBUG] switch_core_state_machine.c:341 (
> sofia/internal/1005 at 10.10.10.16) State ROUTING going to sleep
> 2011-06-12 10:57:57.607036 [DEBUG] switch_core_state_machine.c:314 (
> sofia/internal/1005 at 10.10.10.16) Running State Change CS_CONSUME_MEDIA
> 2011-06-12 10:57:57.607036 [DEBUG] switch_core_state_machine.c:360 (
> sofia/internal/1005 at 10.10.10.16) State CONSUME_MEDIA
> 2011-06-12 10:57:57.607036 [DEBUG] switch_core_state_machine.c:360 (
> sofia/internal/1005 at 10.10.10.16) State CONSUME_MEDIA going to sleep
> 2011-06-12 10:57:57.608051 [DEBUG] sofia.c:4153 Channel
> sofia/internal/1005 at 10.10.10.16 entering state [calling][0]
> 2011-06-12 10:57:57.608051 [DEBUG] sofia.c:5847 IP 10.10.10.16 Rejected by
> acl "domains". Falling back to Digest auth.
> 2011-06-12 10:57:57.608051 [NOTICE] switch_channel.c:669 New Channel
> sofia/internal/1234567890 at 10.10.10.16[86e5b4a9-8c63-4d22-b087-94c4b95d1abb]
> 2011-06-12 10:57:57.609139 [DEBUG] switch_core_state_machine.c:314 (
> sofia/internal/1234567890 at 10.10.10.16) Running State Change CS_NEW
> 2011-06-12 10:57:57.609139 [DEBUG] switch_core_state_machine.c:320 (
> sofia/internal/1234567890 at 10.10.10.16) State NEW
> 2011-06-12 10:57:57.616761 [DEBUG] sofia.c:4153 Channel
> sofia/internal/1234567890 at 10.10.10.16 entering state [received][100]
> 2011-06-12 10:57:57.616761 [DEBUG] sofia.c:4164 Remote SDP:
> v=0
> o=FreeSWITCH 1307870225 1307870226 IN IP4 10.10.10.16
> s=FreeSWITCH
> c=IN IP4 10.10.10.16
> t=0 0
> m=audio 20452 RTP/AVP 115 107 9 0 8 3 101 13
> a=rtpmap:115 G7221/32000
> a=fmtp:115 bitrate=48000
> a=rtpmap:107 G7221/16000
> a=fmtp:107 bitrate=32000
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
> 2011-06-12 10:57:57.616761 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
> [G7221:115:32000:20]/[G7221:115:32000:20]
> 2011-06-12 10:57:57.616761 [DEBUG] sofia_glue.c:2354 Set Codec
> sofia/internal/1234567890 at 10.10.10.16 G7221/32000 20 ms 640 samples
> 2011-06-12 10:57:57.617820 [DEBUG] sofia_glue.c:3524 Set 2833 dtmf
> send/recv payload to 101
> 2011-06-12 10:57:57.617820 [DEBUG] sofia.c:4310 (
> sofia/internal/1234567890 at 10.10.10.16) State Change CS_NEW -> CS_INIT
> 2011-06-12 10:57:57.617820 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/1234567890 at 10.10.10.16 [BREAK]
> 2011-06-12 10:57:57.618925 [DEBUG] switch_core_state_machine.c:314 (
> sofia/internal/1234567890 at 10.10.10.16) Running State Change CS_INIT
> 2011-06-12 10:57:57.618925 [DEBUG] switch_core_state_machine.c:338 (
> sofia/internal/1234567890 at 10.10.10.16) State INIT
> 2011-06-12 10:57:57.618925 [DEBUG] mod_sofia.c:83
> sofia/internal/1234567890 at 10.10.10.16 SOFIA INIT
> 2011-06-12 10:57:57.618925 [DEBUG] mod_sofia.c:117 (
> sofia/internal/1234567890 at 10.10.10.16) State Change CS_INIT -> CS_ROUTING
> 2011-06-12 10:57:57.618925 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/1234567890 at 10.10.10.16 [BREAK]
> 2011-06-12 10:57:57.618925 [DEBUG] switch_core_state_machine.c:338 (
> sofia/internal/1234567890 at 10.10.10.16) State INIT going to sleep
> 2011-06-12 10:57:57.618925 [DEBUG] switch_core_state_machine.c:314 (
> sofia/internal/1234567890 at 10.10.10.16) Running State Change CS_ROUTING
> 2011-06-12 10:57:57.618925 [DEBUG] switch_core_state_machine.c:341 (
> sofia/internal/1234567890 at 10.10.10.16) State ROUTING
> 2011-06-12 10:57:57.618925 [DEBUG] mod_sofia.c:140
> sofia/internal/1234567890 at 10.10.10.16 SOFIA ROUTING
> 2011-06-12 10:57:57.618925 [DEBUG] switch_core_state_machine.c:77
> sofia/internal/1234567890 at 10.10.10.16 Standard ROUTING
> 2011-06-12 10:57:57.618925 [INFO] mod_dialplan_xml.c:418 Processing
> FreeSWITCH_Conference->1005 in context public
> Dialplan: sofia/internal/1234567890 at 10.10.10.16 parsing [public->unloop]
> continue=false
> Dialplan: sofia/internal/1234567890 at 10.10.10.16 Regex (PASS) [unloop]
> ${unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/internal/1234567890 at 10.10.10.16 Regex (PASS) [unloop]
> ${sip_looped_call}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/internal/1234567890 at 10.10.10.16 Action
> deflect(${destination_number})
> 2011-06-12 10:57:57.620002 [DEBUG] switch_core_state_machine.c:119 (
> sofia/internal/1234567890 at 10.10.10.16) State Change CS_ROUTING ->
> CS_EXECUTE
> 2011-06-12 10:57:57.620002 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/1234567890 at 10.10.10.16 [BREAK]
> 2011-06-12 10:57:57.620002 [DEBUG] switch_core_state_machine.c:341 (
> sofia/internal/1234567890 at 10.10.10.16) State ROUTING going to sleep
> 2011-06-12 10:57:57.620002 [DEBUG] switch_core_state_machine.c:314 (
> sofia/internal/1234567890 at 10.10.10.16) Running State Change CS_EXECUTE
> 2011-06-12 10:57:57.620002 [DEBUG] switch_core_state_machine.c:348 (
> sofia/internal/1234567890 at 10.10.10.16) State EXECUTE
> 2011-06-12 10:57:57.620002 [DEBUG] mod_sofia.c:226
> sofia/internal/1234567890 at 10.10.10.16 SOFIA EXECUTE
> 2011-06-12 10:57:57.620002 [DEBUG] switch_core_state_machine.c:157
> sofia/internal/1234567890 at 10.10.10.16 Standard EXECUTE
> EXECUTE sofia/internal/1234567890 at 10.10.10.16 deflect(1005)
> 2011-06-12 10:57:57.621014 [DEBUG] sofia.c:5004 Process REFER to [
> 1005 at 10.10.10.16]
> *2011-06-12 10:57:57.621014 [ERR] sofia.c:5366 Cannot Blind Transfer 1
> Legged calls*
> 2011-06-12 10:57:57.622041 [DEBUG] sofia.c:4153 Channel
> sofia/internal/1005 at 10.10.10.16 entering state [terminated][480]
> 2011-06-12 10:57:57.622041 [NOTICE] sofia.c:4789 Hangup
> sofia/internal/1005 at 10.10.10.16 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE]
> 2011-06-12 10:57:57.622041 [DEBUG] switch_channel.c:2102 Send signal
> sofia/internal/1005 at 10.10.10.16 [KILL]
> 2011-06-12 10:57:57.622041 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/1005 at 10.10.10.16 [BREAK]
> 2011-06-12 10:57:57.622041 [DEBUG] switch_core_state_machine.c:314 (
> sofia/internal/1005 at 10.10.10.16) Running State Change CS_HANGUP
> 2011-06-12 10:57:57.623126 [DEBUG] switch_core_state_machine.c:499 (
> sofia/internal/1005 at 10.10.10.16) State HANGUP
> 2011-06-12 10:57:57.623126 [DEBUG] mod_sofia.c:408
> sofia/internal/1005 at 10.10.10.16 Overriding SIP cause 408 with 480 from the
> other leg
> 2011-06-12 10:57:57.623126 [DEBUG] mod_sofia.c:414 Channel
> sofia/internal/1005 at 10.10.10.16 hanging up, cause: NO_USER_RESPONSE
> 2011-06-12 10:57:57.623126 [DEBUG] switch_ivr_originate.c:3228 Originate
> Resulted in Error Cause: 18 [NO_USER_RESPONSE]
> 2011-06-12 10:57:57.636515 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/1005 at 10.10.10.16 Standard HANGUP, cause: NO_USER_RESPONSE
> 2011-06-12 10:57:57.636515 [DEBUG] switch_core_state_machine.c:499 (
> sofia/internal/1005 at 10.10.10.16) State HANGUP going to sleep
> 2011-06-12 10:57:57.636515 [DEBUG] switch_core_state_machine.c:333 (
> sofia/internal/1005 at 10.10.10.16) State Change CS_HANGUP -> CS_REPORTING
> 2011-06-12 10:57:57.636515 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/1005 at 10.10.10.16 [BREAK]
> 2011-06-12 10:57:57.637437 [DEBUG] switch_core_state_machine.c:314 (
> sofia/internal/1005 at 10.10.10.16) Running State Change CS_REPORTING
> 2011-06-12 10:57:57.637437 [DEBUG] switch_core_state_machine.c:590 (
> sofia/internal/1005 at 10.10.10.16) State REPORTING
> 2011-06-12 10:57:57.637437 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/1005 at 10.10.10.16 Standard REPORTING, cause:
> NO_USER_RESPONSE
> 2011-06-12 10:57:57.637437 [DEBUG] switch_core_state_machine.c:590 (
> sofia/internal/1005 at 10.10.10.16) State REPORTING going to sleep
> 2011-06-12 10:57:57.637437 [DEBUG] switch_core_state_machine.c:327 (
> sofia/internal/1005 at 10.10.10.16) State Change CS_REPORTING -> CS_DESTROY
> 2011-06-12 10:57:57.637437 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/1005 at 10.10.10.16 [BREAK]
> 2011-06-12 10:57:57.637437 [DEBUG] switch_core_session.c:1164 Session 24 (
> sofia/internal/1005 at 10.10.10.16) Locked, Waiting on external entities
> 2011-06-12 10:57:57.637437 [NOTICE] switch_core_session.c:1182 Session 24 (
> sofia/internal/1005 at 10.10.10.16) Ended
> 2011-06-12 10:57:57.637437 [NOTICE] switch_core_session.c:1184 Close
> Channel sofia/internal/1005 at 10.10.10.16 [CS_DESTROY]
> 2011-06-12 10:57:57.637437 [DEBUG] switch_core_state_machine.c:428 (
> sofia/internal/1005 at 10.10.10.16) Running State Change CS_DESTROY
> 2011-06-12 10:57:57.637437 [DEBUG] switch_core_state_machine.c:439 (
> sofia/internal/1005 at 10.10.10.16) State DESTROY
> 2011-06-12 10:57:57.637437 [DEBUG] mod_sofia.c:341
> sofia/internal/1005 at 10.10.10.16 SOFIA DESTROY
> 2011-06-12 10:57:57.637437 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/1005 at 10.10.10.16 Standard DESTROY
> 2011-06-12 10:57:57.637437 [DEBUG] switch_core_state_machine.c:439 (
> sofia/internal/1005 at 10.10.10.16) State DESTROY going to sleep
> 2011-06-12 10:57:57.720295 [DEBUG] switch_core_session.c:641 Send signal
> sofia/internal/1234567890 at 10.10.10.16 [BREAK]
> 2011-06-12 10:57:57.721283 [NOTICE] switch_core_state_machine.c:185
> sofia/internal/1234567890 at 10.10.10.16 has executed the last dialplan
> instruction, hanging up.
> 2011-06-12 10:57:57.721283 [NOTICE] switch_core_state_machine.c:187 Hangup
> sofia/internal/1234567890 at 10.10.10.16 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-06-12 10:57:57.721283 [DEBUG] switch_channel.c:2102 Send signal
> sofia/internal/1234567890 at 10.10.10.16 [KILL]
> 2011-06-12 10:57:57.722210 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/1234567890 at 10.10.10.16 [BREAK]
> 2011-06-12 10:57:57.722210 [DEBUG] switch_core_state_machine.c:348 (
> sofia/internal/1234567890 at 10.10.10.16) State EXECUTE going to sleep
> 2011-06-12 10:57:57.722210 [DEBUG] switch_core_state_machine.c:314 (
> sofia/internal/1234567890 at 10.10.10.16) Running State Change CS_HANGUP
> 2011-06-12 10:57:57.722210 [DEBUG] switch_core_state_machine.c:499 (
> sofia/internal/1234567890 at 10.10.10.16) State HANGUP
> 2011-06-12 10:57:57.722210 [DEBUG] mod_sofia.c:414 Channel
> sofia/internal/1234567890 at 10.10.10.16 hanging up, cause: NORMAL_CLEARING
> 2011-06-12 10:57:57.722210 [DEBUG] mod_sofia.c:476 Responding to INVITE
> with: 480
> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/1234567890 at 10.10.10.16 Standard HANGUP, cause:
> NORMAL_CLEARING
> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:499 (
> sofia/internal/1234567890 at 10.10.10.16) State HANGUP going to sleep
> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:333 (
> sofia/internal/1234567890 at 10.10.10.16) State Change CS_HANGUP ->
> CS_REPORTING
> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/1234567890 at 10.10.10.16 [BREAK]
> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:314 (
> sofia/internal/1234567890 at 10.10.10.16) Running State Change CS_REPORTING
> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:590 (
> sofia/internal/1234567890 at 10.10.10.16) State REPORTING
> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/1234567890 at 10.10.10.16 Standard REPORTING, cause:
> NORMAL_CLEARING
> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:590 (
> sofia/internal/1234567890 at 10.10.10.16) State REPORTING going to sleep
> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:327 (
> sofia/internal/1234567890 at 10.10.10.16) State Change CS_REPORTING ->
> CS_DESTROY
> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/1234567890 at 10.10.10.16 [BREAK]
> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_session.c:1164 Session 25 (
> sofia/internal/1234567890 at 10.10.10.16) Locked, Waiting on external
> entities
> 2011-06-12 10:57:57.723321 [NOTICE] switch_core_session.c:1182 Session 25 (
> sofia/internal/1234567890 at 10.10.10.16) Ended
> 2011-06-12 10:57:57.723321 [NOTICE] switch_core_session.c:1184 Close
> Channel sofia/internal/1234567890 at 10.10.10.16 [CS_DESTROY]
> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:428 (
> sofia/internal/1234567890 at 10.10.10.16) Running State Change CS_DESTROY
> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:439 (
> sofia/internal/1234567890 at 10.10.10.16) State DESTROY
> 2011-06-12 10:57:57.723321 [DEBUG] mod_sofia.c:341
> sofia/internal/1234567890 at 10.10.10.16 SOFIA DESTROY
> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/1234567890 at 10.10.10.16 Standard DESTROY
> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:439 (
> sofia/internal/1234567890 at 10.10.10.16) State DESTROY going to sleep
>
> ##################################################################################
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Jun 13 19:30:26 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 13 Jun 2011 08:30:26 -0700
Subject: [Freeswitch-users] X-Lite giving error while connecting to
 FreeSwitch Server
In-Reply-To: <BANLkTimf-E48+fCz7tUceRkR1zP+HvSXzw@mail.gmail.com>
References: <BANLkTimf-E48+fCz7tUceRkR1zP+HvSXzw@mail.gmail.com>
Message-ID: <BANLkTine-4jZyVgnpVwwDm4p1CRA1AB-0A@mail.gmail.com>

Are you running FreeSWITCH on the same server where you have X-Lite running?
If so, change the host from "localhost" to the actual IP address and try
again.

-MC

On Mon, Jun 13, 2011 at 4:08 AM, ankIT WALiA <ankitwalia4u at gmail.com> wrote:

> Hello all,
>
> I am very new to FreeSwitch and VOIP Technology.
>
> I installed the FreeSwitch and X-Lite on my Window 7 OS.
>
> I am using default FreeSwitch configuration. Nothing changed except adding
> Text to Speech module.
>
> When I am trying connecting to FreeSwitch Sever on localhost from X-Lite
> Softphone. I am getting the error on X-lite
> Account failed to enable.
> Account 1001 could not be enabled.
> Problem at server(SIP error 503). Try Again Later
>
> I have given the account setting at X-Lite as below:
> Account Name: 1001
> Protocol: SIP
> UserID: 1001
> Domain: localhost
> Password: 1234
> Display Name: Test User
> Authorization Name: 1001
>
> Domain Proxy:
> Register with Domain and receive calls: Checked
> Send Outbound via: DOmain
>
> I dont know what wrong I am doing and How I should debug this issue.
>
> Please help.
>
> Thanks
> Ankit
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Jun 13 19:34:25 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 13 Jun 2011 08:34:25 -0700
Subject: [Freeswitch-users] start_dtmf_generate during early media?
In-Reply-To: <1QVra4-0008Vr-8x@mail.aastral.net>
References: <1QVra4-0008Vr-8x@mail.aastral.net>
Message-ID: <BANLkTim5idCYjZLmkhTbXV_dquupQZZd7w@mail.gmail.com>

AFAIK the start_dtmf will work as long as media is established - even early
media.
-MC

On Sun, Jun 12, 2011 at 1:48 PM, Bill W. <freeswitch at aastral.net> wrote:

> Hey everyone,
>
> I'm trying to terminate a call over a voip carrier to an AT&T conference
> bridge.
>
> AT&T (who, of all people, should know better) are playing the prompts
> for the conference bridge over early media.  And of course, you can't
> connect to the conf. bridge because no DTMF is transferred (because the
> call isn't answered yet).
>
> So the question is, is there a way to force FreeSWITCH to transcode 2833
> DTMF from the A-leg and send inband DTMF on the B-leg while the B-leg is
> in early-media state?
>
> I see there is the start_dtmf_generate application, but will this work
> if the b-leg is still in the early-media state?
>
> Thanks!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From revmichael at bethelightchapel.com  Mon Jun 13 19:34:58 2011
From: revmichael at bethelightchapel.com (Rev Michael Carbone)
Date: Mon, 13 Jun 2011 11:34:58 -0400
Subject: [Freeswitch-users] Timer Config Off
Message-ID: <20110613113458.19515fiw9kw6axgc@psychicawakeningschool.com>

Not sure what happened but recording a conference was working, now I  
get this error when I try to record a conference:

2011-06-13 15:33:16.337400 [ERR] mod_conference.c:2897 Error Opening  
File  
[recordings/conference_CallInLineRadio-localhost-2011-06-13-06_3.wav]

2011-06-13 15:33:16.337400 [ERR] switch_core_timer.c:117 Timer is not  
properly configured.


any ideas?



From Hector.Geraldino at ip-soft.net  Mon Jun 13 20:37:33 2011
From: Hector.Geraldino at ip-soft.net (Hector Geraldino)
Date: Mon, 13 Jun 2011 12:37:33 -0400
Subject: [Freeswitch-users] originating a call from an outbound
	socket	app
In-Reply-To: <BANLkTimhqY1SuQMeWFm1mv=CvQXm72RbEw@mail.gmail.com>
References: <6A6B4C284AD15042B429EB9D904544AD021F45EAA0@NY1-EXMB-01.ip-soft.net>
	<BANLkTi=LX=cGqSVrT6o-0OOPXPOA2U=MnA@mail.gmail.com>
	<6A6B4C284AD15042B429EB9D904544AD021F45EB71@NY1-EXMB-01.ip-soft.net>
	<BANLkTikG8TQ+1JhAVr8Y6xMWhH1xqor8zA@mail.gmail.com>
	<6A6B4C284AD15042B429EB9D904544AD021F45EBBC@NY1-EXMB-01.ip-soft.net>
	<BANLkTimhqY1SuQMeWFm1mv=CvQXm72RbEw@mail.gmail.com>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD021F45EC87@NY1-EXMB-01.ip-soft.net>

Hi Michael,

I finally figure out how to connect both calls using the command 'intercept'. From the inbound call (handled by the OB socket) I just do an execute intercept legB uuid, and both calls are bridged perfectly.  Thanks very much.

So the only thing that stills concerns me is why the event CHANNEL_ANSWER is always generated on the outbound call (legb), even if the user doesn't pick up the call. Also I can't find any differences on the CHANNEL_CALLSTATE or CALL_UPDATE events that allows me to determine when an outbound call generated from FS is actually answered or not.

BTW, I bought the FS 1.0.6 book on Amazon last week and I want to congratulate you and Anthony/Darren for the excellent work you've done, it has been really helpful.  My only complaint is that it doesn't have an eBook format so I can read it on my Kindle. Besides that, great work.


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Friday, June 10, 2011 6:27 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] originating a call from an outbound socket app


On Fri, Jun 10, 2011 at 7:00 AM, Hector Geraldino <Hector.Geraldino at ip-soft.net<mailto:Hector.Geraldino at ip-soft.net>> wrote:
Thanks for your response Michael.

Finally I found the way of dialing to an external extension. What I did was just open a shell, listen in a port using nc  and tried to originate a call. It worked using the syntax:

originate sofia/internal/5562 at 192.168.23.104<mailto:5562 at 192.168.23.104> &park

As this is an outbound call (not managed by the socket application) I had to use a new inbound socket connection to connect to FS, make the call, query for the uuid and receive some events (like DTMF among others).  Everything was going great, until I found two issues:


-          The CHANNEL_ANSWER event is always triggered, no matter if the call is actually picked up or not on my phone. I thought that maybe I can use some other event to determine if the call have been answered or not (like CHANNEL_CALLSTATE or CALL_UPDATE), but the additional information on the event headers is the same no matter if it's answered or not.

-          When the call is answered, and the user presses a key to accept the call, the function returns the UUID of this session to the main method. Then I tried to make the bridge between the two calls (the legA call handled by the socket application, and the legB outbound call generated in the application using an inbound connection). I using the command

api uuid_bridge <incoming_uuid> <outbound_uuid>

but this didn't work. Am I missing something here?
You're missing the debug log on the pastebin. ;)

Yeah, let's take a look at the debug log of this happening. Capture from the beginning of the a leg all the way through to trying the uuid_bridge.
-MC

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From ankitwalia4u at gmail.com  Mon Jun 13 21:45:33 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Mon, 13 Jun 2011 23:15:33 +0530
Subject: [Freeswitch-users] X-Lite giving error while connecting to
 FreeSwitch Server
In-Reply-To: <BANLkTine-4jZyVgnpVwwDm4p1CRA1AB-0A@mail.gmail.com>
References: <BANLkTimf-E48+fCz7tUceRkR1zP+HvSXzw@mail.gmail.com>
	<BANLkTine-4jZyVgnpVwwDm4p1CRA1AB-0A@mail.gmail.com>
Message-ID: <BANLkTimLrgSK+=fFH6cRRbRgXMU36OMdPg@mail.gmail.com>

I tried your suggestion and it is working now, Thank you very much.

On Mon, Jun 13, 2011 at 9:00 PM, Michael Collins <msc at freeswitch.org> wrote:

> Are you running FreeSWITCH on the same server where you have X-Lite
> running? If so, change the host from "localhost" to the actual IP address
> and try again.
>
> -MC
>
> On Mon, Jun 13, 2011 at 4:08 AM, ankIT WALiA <ankitwalia4u at gmail.com>wrote:
>
>> Hello all,
>>
>> I am very new to FreeSwitch and VOIP Technology.
>>
>> I installed the FreeSwitch and X-Lite on my Window 7 OS.
>>
>> I am using default FreeSwitch configuration. Nothing changed except adding
>> Text to Speech module.
>>
>> When I am trying connecting to FreeSwitch Sever on localhost from X-Lite
>> Softphone. I am getting the error on X-lite
>> Account failed to enable.
>> Account 1001 could not be enabled.
>> Problem at server(SIP error 503). Try Again Later
>>
>> I have given the account setting at X-Lite as below:
>> Account Name: 1001
>> Protocol: SIP
>> UserID: 1001
>> Domain: localhost
>> Password: 1234
>> Display Name: Test User
>> Authorization Name: 1001
>>
>> Domain Proxy:
>> Register with Domain and receive calls: Checked
>> Send Outbound via: DOmain
>>
>> I dont know what wrong I am doing and How I should debug this issue.
>>
>> Please help.
>>
>> Thanks
>> Ankit
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Life is like a rose its upto u feel it as its fragrance or thorns
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From cbanta at gmail.com  Mon Jun 13 20:31:47 2011
From: cbanta at gmail.com (Cameron Banta)
Date: Mon, 13 Jun 2011 11:31:47 -0500
Subject: [Freeswitch-users] conference high cpu and audio delay
Message-ID: <4DF63B73.3090902@gmail.com>

Hello,

I'm running FreeSWITCH Version 1.0.7 (hacked-20110606T040042Z) on Ubuntu 
10.04 64 bit with kernel 2.6.35-22-server.

I am running about 300 simultaneous calls spread out over about 50 
conferences. I'm getting a cpu load of 20 to 30, and cpu utilization 
about 75% on each core. There's about a 2 second audio delay in the 
conference when it gets this loaded.

Is this high load and audio delay just the limit of my cpu (dual core 
E5200)?

I ask because I was able to get the same number of calls on this server 
with asterisk 1.6 with app_konference with a much lower cpu load - about 
4 or 5 - and no audio delay.

Thanks,
Cameron


From yungwei at resolvity.com  Mon Jun 13 21:49:16 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Mon, 13 Jun 2011 13:49:16 -0400
Subject: [Freeswitch-users] originate a call to an extension and then
 connect to another extension
Message-ID: <33095823FD21DF429B481B5163264B7950AC643195@VMBX102.ihostexchange.net>

Hi,

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


From msc at freeswitch.org  Mon Jun 13 21:52:57 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 13 Jun 2011 10:52:57 -0700
Subject: [Freeswitch-users] originating a call from an outbound socket
	app
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD021F45EC87@NY1-EXMB-01.ip-soft.net>
References: <6A6B4C284AD15042B429EB9D904544AD021F45EAA0@NY1-EXMB-01.ip-soft.net>
	<BANLkTi=LX=cGqSVrT6o-0OOPXPOA2U=MnA@mail.gmail.com>
	<6A6B4C284AD15042B429EB9D904544AD021F45EB71@NY1-EXMB-01.ip-soft.net>
	<BANLkTikG8TQ+1JhAVr8Y6xMWhH1xqor8zA@mail.gmail.com>
	<6A6B4C284AD15042B429EB9D904544AD021F45EBBC@NY1-EXMB-01.ip-soft.net>
	<BANLkTimhqY1SuQMeWFm1mv=CvQXm72RbEw@mail.gmail.com>
	<6A6B4C284AD15042B429EB9D904544AD021F45EC87@NY1-EXMB-01.ip-soft.net>
Message-ID: <BANLkTinZP-2TY9QTsjCX7D2cO4wj7mA-Pg@mail.gmail.com>

On Mon, Jun 13, 2011 at 9:37 AM, Hector Geraldino <
Hector.Geraldino at ip-soft.net> wrote:

> Hi Michael,
>
>
>
> I finally figure out how to connect both calls using the command
> ?intercept?. From the inbound call (handled by the OB socket) I just do an
> execute intercept legB uuid, and both calls are bridged perfectly.  Thanks
> very much.
>
>
>
> So the only thing that stills concerns me is why the event CHANNEL_ANSWER
> is always generated on the outbound call (legb), even if the user doesn?t
> pick up the call. Also I can?t find any differences on the CHANNEL_CALLSTATE
> or CALL_UPDATE events that allows me to determine when an outbound call
> generated from FS is actually answered or not.
>
Did you ever post the full call debug log with a siptrace? I don't recall
seeing that...


>
>
> BTW, I bought the FS 1.0.6 book on Amazon last week and I want to
> congratulate you and Anthony/Darren for the excellent work you?ve done, it
> has been really helpful.  My only complaint is that it doesn?t have an eBook
> format so I can read it on my Kindle. Besides that, great work.
>
Yes, that complaint has been levied on more than one occasion. We've been
imploring Packt to rectify the situation. We'll keep you posted on the
status.

-MC
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From freeswitch at aastral.net  Mon Jun 13 22:00:37 2011
From: freeswitch at aastral.net (Bill W.)
Date: Mon, 13 Jun 2011 14:00:37 -0400
Subject: [Freeswitch-users] start_dtmf_generate during early media?
In-Reply-To: <BANLkTim5idCYjZLmkhTbXV_dquupQZZd7w@mail.gmail.com>
References: <1QVra4-0008Vr-8x@mail.aastral.net>
	<BANLkTim5idCYjZLmkhTbXV_dquupQZZd7w@mail.gmail.com>
Message-ID: <1QWBRK-0007Rl-Qf@mail.aastral.net>

Hey Michael,

Thanks for the reply! If that is the case, what is the best way to
trigger that?

I've tried setting this on the A-leg, but it didn't appear to work.

<action application="export"
data="nolocal:execute_on_media=start_dtmf_generate true"/>

Thanks,
Bill


On 6/13/11 11:34 AM, Michael Collins wrote:
> AFAIK the start_dtmf will work as long as media is established - even
> early media.
> -MC


From errotan at elder.hu  Mon Jun 13 22:27:36 2011
From: errotan at elder.hu (=?iso-8859-1?q?Pusk=E1s_Zsolt?=)
Date: Mon, 13 Jun 2011 20:27:36 +0200
Subject: [Freeswitch-users] conference high cpu and audio delay
In-Reply-To: <4DF63B73.3090902@gmail.com>
References: <4DF63B73.3090902@gmail.com>
Message-ID: <201106132027.36630.errotan@elder.hu>

Hi.

You should try to compile an 1000 Hz kernel for your box. I helps alot.


2011. j?nius 13. 18:31:47 d?tummal Cameron Banta az al?bbiakat ?rta:
> Hello,
> 
> I'm running FreeSWITCH Version 1.0.7 (hacked-20110606T040042Z) on Ubuntu
> 10.04 64 bit with kernel 2.6.35-22-server.
> 
> I am running about 300 simultaneous calls spread out over about 50
> conferences. I'm getting a cpu load of 20 to 30, and cpu utilization
> about 75% on each core. There's about a 2 second audio delay in the
> conference when it gets this loaded.
> 
> Is this high load and audio delay just the limit of my cpu (dual core
> E5200)?
> 
> I ask because I was able to get the same number of calls on this server
> with asterisk 1.6 with app_konference with a much lower cpu load - about
> 4 or 5 - and no audio delay.
> 
> Thanks,
> Cameron
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From msc at freeswitch.org  Mon Jun 13 22:32:53 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 13 Jun 2011 11:32:53 -0700
Subject: [Freeswitch-users] start_dtmf_generate during early media?
In-Reply-To: <1QWBRK-0007Rl-Qf@mail.aastral.net>
References: <1QVra4-0008Vr-8x@mail.aastral.net>
	<BANLkTim5idCYjZLmkhTbXV_dquupQZZd7w@mail.gmail.com>
	<1QWBRK-0007Rl-Qf@mail.aastral.net>
Message-ID: <BANLkTikge0n3rxzPu8RY5ZNAbTVad3NUQg@mail.gmail.com>

Can you pb the debug output of this call? You might also record the call and
listen to the audio, just to confirm what's happening.

-MC

On Mon, Jun 13, 2011 at 11:00 AM, Bill W. <freeswitch at aastral.net> wrote:

> Hey Michael,
>
> Thanks for the reply! If that is the case, what is the best way to
> trigger that?
>
> I've tried setting this on the A-leg, but it didn't appear to work.
>
> <action application="export"
> data="nolocal:execute_on_media=start_dtmf_generate true"/>
>
> Thanks,
> Bill
>
>
> On 6/13/11 11:34 AM, Michael Collins wrote:
> > AFAIK the start_dtmf will work as long as media is established - even
> > early media.
> > -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Jun 13 22:36:53 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 13 Jun 2011 11:36:53 -0700
Subject: [Freeswitch-users] originate a call to an extension and then
 connect to another extension
In-Reply-To: <33095823FD21DF429B481B5163264B7950AC643195@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950AC643195@VMBX102.ihostexchange.net>
Message-ID: <BANLkTinnUwoN3nrMy0gioUmpyzh1LNg_tg@mail.gmail.com>

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

originate user/1000 1001

If you want to call 1001 first then do this:

originate user/1001 1000

-MC

On Mon, Jun 13, 2011 at 10:49 AM, Yungwei Chen <yungwei at resolvity.com>wrote:

> Hi,
>
> I am wondering if it's possible to originate a call to an extension and
> then connect to another extension. If so, how does the the dialstring look
> like? Thanks.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Hector.Geraldino at ip-soft.net  Mon Jun 13 22:47:55 2011
From: Hector.Geraldino at ip-soft.net (Hector Geraldino)
Date: Mon, 13 Jun 2011 14:47:55 -0400
Subject: [Freeswitch-users] originating a call from an outbound
	socket	app
In-Reply-To: <BANLkTinZP-2TY9QTsjCX7D2cO4wj7mA-Pg@mail.gmail.com>
References: <6A6B4C284AD15042B429EB9D904544AD021F45EAA0@NY1-EXMB-01.ip-soft.net>
	<BANLkTi=LX=cGqSVrT6o-0OOPXPOA2U=MnA@mail.gmail.com>
	<6A6B4C284AD15042B429EB9D904544AD021F45EB71@NY1-EXMB-01.ip-soft.net>
	<BANLkTikG8TQ+1JhAVr8Y6xMWhH1xqor8zA@mail.gmail.com>
	<6A6B4C284AD15042B429EB9D904544AD021F45EBBC@NY1-EXMB-01.ip-soft.net>
	<BANLkTimhqY1SuQMeWFm1mv=CvQXm72RbEw@mail.gmail.com>
	<6A6B4C284AD15042B429EB9D904544AD021F45EC87@NY1-EXMB-01.ip-soft.net>
	<BANLkTinZP-2TY9QTsjCX7D2cO4wj7mA-Pg@mail.gmail.com>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD021F45ECAA@NY1-EXMB-01.ip-soft.net>

Right, the call log is on:

http://pastebin.com/kAEZsFe6

I found the issue which is really silly (don't know how I couldn't figure it out before): the call has been picked up by the voicemail, that's why the CHANNEL_ANSWER event is shown up.

What's the most common pattern to deal with this situation, when you want to know if the call has been answered by a human being and not by a machine? I'm thinking about requesting a confirmation (DTMF) and timeout the operation after a few secs, but as this seems to be a very common task, I was wondering if there's a better way of doing it.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Monday, June 13, 2011 1:53 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] originating a call from an outbound socket app


On Mon, Jun 13, 2011 at 9:37 AM, Hector Geraldino <Hector.Geraldino at ip-soft.net<mailto:Hector.Geraldino at ip-soft.net>> wrote:
Hi Michael,

I finally figure out how to connect both calls using the command 'intercept'. From the inbound call (handled by the OB socket) I just do an execute intercept legB uuid, and both calls are bridged perfectly.  Thanks very much.

So the only thing that stills concerns me is why the event CHANNEL_ANSWER is always generated on the outbound call (legb), even if the user doesn't pick up the call. Also I can't find any differences on the CHANNEL_CALLSTATE or CALL_UPDATE events that allows me to determine when an outbound call generated from FS is actually answered or not.
Did you ever post the full call debug log with a siptrace? I don't recall seeing that...


BTW, I bought the FS 1.0.6 book on Amazon last week and I want to congratulate you and Anthony/Darren for the excellent work you've done, it has been really helpful.  My only complaint is that it doesn't have an eBook format so I can read it on my Kindle. Besides that, great work.
Yes, that complaint has been levied on more than one occasion. We've been imploring Packt to rectify the situation. We'll keep you posted on the status.

-MC

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From yungwei at resolvity.com  Mon Jun 13 23:08:04 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Mon, 13 Jun 2011 15:08:04 -0400
Subject: [Freeswitch-users] originate a call to an extension and then
 connect to another extension
In-Reply-To: <BANLkTinnUwoN3nrMy0gioUmpyzh1LNg_tg@mail.gmail.com>
References: <33095823FD21DF429B481B5163264B7950AC643195@VMBX102.ihostexchange.net>
	<BANLkTinnUwoN3nrMy0gioUmpyzh1LNg_tg@mail.gmail.com>
Message-ID: <33095823FD21DF429B481B5163264B7950AC6431DC@VMBX102.ihostexchange.net>

I don't mean registered users. I mean extensions defined in conf/dialplan/default/my_exten.xml as shown below.
In this case, I can do: originate sofia/internal/answer question?

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

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

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

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

originate user/1000 1001

If you want to call 1001 first then do this:

originate user/1001 1000

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

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

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

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From msc at freeswitch.org  Tue Jun 14 00:59:21 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 13 Jun 2011 13:59:21 -0700
Subject: [Freeswitch-users] originate a call to an extension and then
 connect to another extension
In-Reply-To: <33095823FD21DF429B481B5163264B7950AC6431DC@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950AC643195@VMBX102.ihostexchange.net>
	<BANLkTinnUwoN3nrMy0gioUmpyzh1LNg_tg@mail.gmail.com>
	<33095823FD21DF429B481B5163264B7950AC6431DC@VMBX102.ihostexchange.net>
Message-ID: <BANLkTiki1Cd6Z05UfmF+JQA1hdx754ngMw@mail.gmail.com>

originate loopback/question answer

On Mon, Jun 13, 2011 at 12:08 PM, Yungwei Chen <yungwei at resolvity.com>wrote:

> I don't mean registered users. I mean extensions defined in
> conf/dialplan/default/my_exten.xml as shown below.
>
> In this case, I can do: originate sofia/internal/answer question?
>
>
>
>   <extension name="question">
>
>       <condition field="destination_number" expression="^questiony$">
>
>         <action application="answer" />
>
>         <action application="playback"
> data="/usr/local/freeswitch/sounds/question.wav" />
>
>       </condition>
>
>   </extension>
>
>
>
>   <extension name="answer">
>
>       <condition field="destination_number" expression="^answer$">
>
>         <action application="answer" />
>
>         <action application="playback"
> data="/usr/local/freeswitch/sounds/answer.wav" />
>
>       </condition>
>
>   </extension>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Monday, June 13, 2011 1:37 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] originate a call to an extension and
> then connect to another extension
>
>
>
> If you mean you want to call x1000 and then when it answers, immediately
> dial x1001? If so it's just this, assuming these are registered users:
>
>
>
> originate user/1000 1001
>
>
>
> If you want to call 1001 first then do this:
>
>
>
> originate user/1001 1000
>
>
>
> -MC
>
> On Mon, Jun 13, 2011 at 10:49 AM, Yungwei Chen wrote:
>
> Hi,
>
> I am wondering if it's possible to originate a call to an extension and
> then connect to another extension. If so, how does the the dialstring look
> like? Thanks.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Jun 14 01:02:16 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 13 Jun 2011 14:02:16 -0700
Subject: [Freeswitch-users] originating a call from an outbound socket
	app
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD021F45ECAA@NY1-EXMB-01.ip-soft.net>
References: <6A6B4C284AD15042B429EB9D904544AD021F45EAA0@NY1-EXMB-01.ip-soft.net>
	<BANLkTi=LX=cGqSVrT6o-0OOPXPOA2U=MnA@mail.gmail.com>
	<6A6B4C284AD15042B429EB9D904544AD021F45EB71@NY1-EXMB-01.ip-soft.net>
	<BANLkTikG8TQ+1JhAVr8Y6xMWhH1xqor8zA@mail.gmail.com>
	<6A6B4C284AD15042B429EB9D904544AD021F45EBBC@NY1-EXMB-01.ip-soft.net>
	<BANLkTimhqY1SuQMeWFm1mv=CvQXm72RbEw@mail.gmail.com>
	<6A6B4C284AD15042B429EB9D904544AD021F45EC87@NY1-EXMB-01.ip-soft.net>
	<BANLkTinZP-2TY9QTsjCX7D2cO4wj7mA-Pg@mail.gmail.com>
	<6A6B4C284AD15042B429EB9D904544AD021F45ECAA@NY1-EXMB-01.ip-soft.net>
Message-ID: <BANLkTikereRiQ_hVpeqh=TO9hRdAWxx=9w@mail.gmail.com>

Well, you can use the answer confirmation variables...
http://wiki.freeswitch.org/wiki/Channel_Variables#Answer_confirmation_variables

or you can contact consulting at freeswitch.org and see about getting the
answering machine detection module. (Not a free item, BTW.)

-MC

On Mon, Jun 13, 2011 at 11:47 AM, Hector Geraldino <
Hector.Geraldino at ip-soft.net> wrote:

> Right, the call log is on:
>
>
>
> http://pastebin.com/kAEZsFe6
>
>
>
> I found the issue which is really silly (don?t know how I couldn?t figure
> it out before): the call has been picked up by the voicemail, that?s why the
> CHANNEL_ANSWER event is shown up.
>
>
>
> What?s the most common pattern to deal with this situation, when you want
> to know if the call has been answered by a human being and not by a machine?
> I?m thinking about requesting a confirmation (DTMF) and timeout the
> operation after a few secs, but as this seems to be a very common task, I
> was wondering if there?s a better way of doing it.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Monday, June 13, 2011 1:53 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] originating a call from an outbound
> socket app
>
>
>
>
>
> On Mon, Jun 13, 2011 at 9:37 AM, Hector Geraldino <
> Hector.Geraldino at ip-soft.net> wrote:
>
> Hi Michael,
>
>
>
> I finally figure out how to connect both calls using the command
> ?intercept?. From the inbound call (handled by the OB socket) I just do an
> execute intercept legB uuid, and both calls are bridged perfectly.  Thanks
> very much.
>
>
>
> So the only thing that stills concerns me is why the event CHANNEL_ANSWER
> is always generated on the outbound call (legb), even if the user doesn?t
> pick up the call. Also I can?t find any differences on the CHANNEL_CALLSTATE
> or CALL_UPDATE events that allows me to determine when an outbound call
> generated from FS is actually answered or not.
>
> Did you ever post the full call debug log with a siptrace? I don't recall
> seeing that...
>
>
>
>
>
> BTW, I bought the FS 1.0.6 book on Amazon last week and I want to
> congratulate you and Anthony/Darren for the excellent work you?ve done, it
> has been really helpful.  My only complaint is that it doesn?t have an eBook
> format so I can read it on my Kindle. Besides that, great work.
>
> Yes, that complaint has been levied on more than one occasion. We've been
> imploring Packt to rectify the situation. We'll keep you posted on the
> status.
>
>
>
> -MC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Jun 14 01:04:29 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 13 Jun 2011 14:04:29 -0700
Subject: [Freeswitch-users] regex groups
In-Reply-To: <assp.0143d10b6f.92097A6A775D5147B1078E3F15430B9266ED53@prato.activenetwerx.local>
References: <assp.0143d10b6f.92097A6A775D5147B1078E3F15430B9266ED53@prato.activenetwerx.local>
Message-ID: <BANLkTikOi-aPic6MtM7ddVzWXTtG=FT+bw@mail.gmail.com>

What's in $0 on something like this?
expresion="^0(\d+)$"

Just curious. This is probably a minor bug, so open a jira w/ low priority.
-MC

On Sat, Jun 11, 2011 at 12:12 PM, Joseph L. Casale <
jcasale at activenetwerx.com> wrote:

> I wrote up a quick dialplan and intended on using $0 for the whole number,
> so my regex was:
>     <condition field="destination_number" expression="^0\d+$">
> but $0 was empty until I rewrote it:
>     <condition field="destination_number" expression="^(0\d+)$">
>
> Is this expected, I assumed not and I was not looking for $1 etc I figured
> $0 to
> always get populated.
>
> Thanks,
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Jun 14 01:08:09 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 13 Jun 2011 14:08:09 -0700
Subject: [Freeswitch-users] CHANNEL_BRIDGE event missing
In-Reply-To: <20110611170858.278240@gmx.com>
References: <20110611170858.278240@gmx.com>
Message-ID: <BANLkTin39vkrAOHWk3uXeD5iCa7H6k6oQQ@mail.gmail.com>

Can you do a few tests? First, filter for just the CHANNEL_BRIDGE events (so
you see them for all channels) and then make your test call. Confirm that
they are coming through before you bang your head any further...

-MC

On Sat, Jun 11, 2011 at 10:08 AM, george russell <georuse at gmx.com> wrote:

> Freeswitch has connected my call. The call has gone through the gateway and
> connected to the phone on the other end.
> Because the call has connected I thought CHANNEL_BRIDGE should of occured.
>
> This is the list of events captured. Why would CHANNEL_BRIDGE event be
> missing even though the call has been bridged?
>
>
> I have done a "events plain all" and filtered for aleg uuid and bleg uuid
>
> "filter Unique-ID (aleg uuid)"
> "filter Unique-ID (bleg uuid)"
>
>
>
> CHANNEL_EXECUTE / export
> CHANNEL_EXECUTE_COMPLETE / export
> CHANNEL_EXECUTE / export
> CHANNEL_EXECUTE_COMPLETE / export
> CHANNEL_EXECUTE / export
> CHANNEL_EXECUTE_COMPLETE / export
> CHANNEL_EXECUTE / export
> CHANNEL_EXECUTE_COMPLETE / export
> CHANNEL_EXECUTE / export
> CHANNEL_EXECUTE_COMPLETE / export
> CHANNEL_EXECUTE / export
> CHANNEL_EXECUTE_COMPLETE / export
> CHANNEL_EXECUTE / export
> CHANNEL_EXECUTE_COMPLETE / export
> CHANNEL_EXECUTE / bridge
> CHANNEL_PROGRESS /
> CALL_UPDATE /
> CALL_UPDATE /
> CALL_UPDATE /
> CALL_UPDATE /
> CHANNEL_ANSWER /
> SESSION_HEARTBEAT /
> CHANNEL_HANGUP /
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Jun 14 01:11:11 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 13 Jun 2011 14:11:11 -0700
Subject: [Freeswitch-users] Lua ASR TTS
In-Reply-To: <1307202350274-6439449.post@n2.nabble.com>
References: <1307202350274-6439449.post@n2.nabble.com>
Message-ID: <BANLkTimUAy28CHKTA4rm_31bA5h+_PBVJw@mail.gmail.com>

Sorry for the late reply - I missed this one. Are you using the break app or
the uuid_break API? Also, are you using the "all" argument?
-MC

On Sat, Jun 4, 2011 at 8:45 AM, educs13 <educs13 at yahoo.com.br> wrote:

> Hi,
>
> I was trying to run the 'Examples directory lua asr tts'
> (http://wiki.freeswitch.org/wiki/Examples_directory_lua_asr_tts) in Ubuntu
> 10.04/32 bits, but I had some problems with the 'session:streamFile()'.
>
> If I say something when the prompt is still playing, the audio stucks
> indefinitely. It seems that the 'break' command isn't doing all its job. I
> guess that maybe the 'break' is just stop feeding the audio player with new
> samples, but it is not stopping the player. So, this player keeps reading
> and playing the same samples continuously ...
>
> I really don't know if my guess makes sense and as I don't know C very
> well,
> I don't know where to look for a solution in the source code. I Hope you
> could help me.
>
> Thanks,
> Eduardo
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Lua-ASR-TTS-tp6439449p6439449.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From david.villasmil.work at gmail.com  Tue Jun 14 02:59:14 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 14 Jun 2011 00:59:14 +0200
Subject: [Freeswitch-users] bypass_media problem
Message-ID: <BANLkTinMGHmdvhrgWXV8OMJadzPSOFcawg@mail.gmail.com>

Hello Guys,

I have a set where i'm receiving a call from a gw and sending to another gw.
Both un-natted. If I set in my (lua) script bypass_media=false, call goes
just fine, if i set it to true, FS sends "Temporary unavailable" to the
B-side and CANCELs the A-side...

Codec is fine as you will see in the pastebin..


here's the trace:

http://pastebin.freeswitch.org/16480

and here's the console log:

http://pastebin.freeswitch.org/16481

I'm bridging with a lua script, this is the pertinent part:

gw_sip_username and gw_sip_pwd are just bogus "1234","1234", no registration
is needed....





 session:execute("set","continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,407")

     session:execute("set","inbound-late-negotiation=true")
     session:execute("set","inbound-bypass-media=false")

     session:execute("set","proxy_media=false")
     session:execute("set","bypass_media=false")


     --session:execute("bridge","{sip_auth_username=" .. gw_sip_username ..
",sip_auth_password=" .. gw_sip_pwd .. "}sofia/external/".. out_number
.."@".. gw_sip_ip .."")


     fsLog("BRIDGE EXECUTE:", "{loop=3}sofia/gateway/${distributor(" ..
route_name .. ")}" .. out_number .. "")
     session:execute("bridge","{loop=3}sofia/gateway/${distributor(" ..
route_name .. ")}".. out_number .."")

     -- hangup
     session:hangup();



Thanks guys for your help!


David
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From david.villasmil.work at gmail.com  Tue Jun 14 03:01:29 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 14 Jun 2011 01:01:29 +0200
Subject: [Freeswitch-users] bypass_media problem
In-Reply-To: <BANLkTinMGHmdvhrgWXV8OMJadzPSOFcawg@mail.gmail.com>
References: <BANLkTinMGHmdvhrgWXV8OMJadzPSOFcawg@mail.gmail.com>
Message-ID: <BANLkTikgJT67eaDUz+0-iYuRbYL0YoV-NQ@mail.gmail.com>

Forgot to mention, calls gets disconnected whe 183 is received from B-side

Thanks!


On Tue, Jun 14, 2011 at 12:59 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello Guys,
>
> I have a set where i'm receiving a call from a gw and sending to another
> gw. Both un-natted. If I set in my (lua) script bypass_media=false, call
> goes just fine, if i set it to true, FS sends "Temporary unavailable" to the
> B-side and CANCELs the A-side...
>
> Codec is fine as you will see in the pastebin..
>
>
> here's the trace:
>
> http://pastebin.freeswitch.org/16480
>
> and here's the console log:
>
> http://pastebin.freeswitch.org/16481
>
> I'm bridging with a lua script, this is the pertinent part:
>
> gw_sip_username and gw_sip_pwd are just bogus "1234","1234", no
> registration is needed....
>
>
>
>
>
>  session:execute("set","continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,407")
>
>      session:execute("set","inbound-late-negotiation=true")
>      session:execute("set","inbound-bypass-media=false")
>
>      session:execute("set","proxy_media=false")
>      session:execute("set","bypass_media=false")
>
>
>      --session:execute("bridge","{sip_auth_username=" .. gw_sip_username ..
> ",sip_auth_password=" .. gw_sip_pwd .. "}sofia/external/".. out_number
> .."@".. gw_sip_ip .."")
>
>
>      fsLog("BRIDGE EXECUTE:", "{loop=3}sofia/gateway/${distributor(" ..
> route_name .. ")}" .. out_number .. "")
>      session:execute("bridge","{loop=3}sofia/gateway/${distributor(" ..
> route_name .. ")}".. out_number .."")
>
>      -- hangup
>      session:hangup();
>
>
>
> Thanks guys for your help!
>
>
> David
>
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From gustavo.espeche at easyipcall.com  Tue Jun 14 03:20:14 2011
From: gustavo.espeche at easyipcall.com (Gustavo Espeche)
Date: Mon, 13 Jun 2011 20:20:14 -0300
Subject: [Freeswitch-users] unload pcmu and pcma
Message-ID: <1308007214.2014.1.camel@gustavo-laptop>

Hi we buy a sangoma d100 and we are having trouble because sangoma
support inform to us that we can't do a trascoding g711-->g729 because
we can unload the codec g711 from freeswitch.
Some one know how we can fix this issue?
Best Regards.

-- 

Gustavo Espeche
EasyIpCall S.R.L.
www.easyipcall.com
Bv Mitre 517 24? E
Cordoba - Argentina
Te: +54 - 351 - 4280633



From curriegrad2004 at gmail.com  Tue Jun 14 03:24:40 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 13 Jun 2011 16:24:40 -0700
Subject: [Freeswitch-users] unload pcmu and pcma
In-Reply-To: <1308007214.2014.1.camel@gustavo-laptop>
References: <1308007214.2014.1.camel@gustavo-laptop>
Message-ID: <BANLkTik92ax0oWiRmJSt=Pc2rW0JiPhcsw@mail.gmail.com>

G711 is built into the core, you can't just unload it. That specific
Sangoma's support personnel might be confused with how FreeSwitch
works. Try pulling the latest git again, I saw some fixes for the
Sangoma cards that somebody put up a few days ago.

On Mon, Jun 13, 2011 at 4:20 PM, Gustavo Espeche
<gustavo.espeche at easyipcall.com> wrote:
> Hi we buy a sangoma d100 and we are having trouble because sangoma
> support inform to us that we can't do a trascoding g711-->g729 because
> we can unload the codec g711 from freeswitch.
> Some one know how we can fix this issue?
> Best Regards.
>
> --
>
> Gustavo Espeche
> EasyIpCall S.R.L.
> www.easyipcall.com
> Bv Mitre 517 24? E
> Cordoba - Argentina
> Te: +54 - 351 - 4280633
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From ivank at rogers.com  Tue Jun 14 04:23:13 2011
From: ivank at rogers.com (Ivan Kovacevic)
Date: Mon, 13 Jun 2011 20:23:13 -0400
Subject: [Freeswitch-users] continue_on_fail failover with mod_lcr
Message-ID: <041201cc2a29$3ffe2df0$bffa89d0$@rogers.com>

Anyone? 

 

I am still wondering if there is a documented/undocumented reason why continue_on_fail does not work with piped failover?

 

 

 

From: Ivan Kovacevic [mailto:ivank at rogers.com] 
Sent: June-09-11 4:38 PM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: continue_on_fail failover with mod_lcr

 

Hi Everyone,

 

I would like to implement fail-over for my outbound gateway, but I would like to be able to pick certain cause codes which qualify
to stop trying next gateway (specifically when I have bad number and I am getting sip:404 NO_ROUTE_DESTINATION, UNALLOCATED_NUMBER).
I was using 1.0.6, but I moved to the newest git about month ago.

 

After searching through lists and spending several hours playing with continue_to_fail, failure_causes and fail_on_single_reject I
was able to make it work by specifying  cause codes for which I want to fail-over and omitting ones that qualify to stop trying next
gateway in variable continue_to_fail:

 

So this setup is working for me and in the case I have bad number and x.x.x.x returns sip:404 (NO_ROUTE_DESTINATION or
UNALLOCATED_NUMBER) it is not trying y.y.y.y gateway.

 

 

<include>

  <extension name="myextension">

    <condition field="destination_number" expression="^(01144\d+)$">

      <action application="set"

data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,SERVICE_NOT_IMPLEMENTED,CALL_RE

JECTED,NO_USER_RESPONSE,EXCHANGE_ROUTING_ERROR,INCOMPATIBLE_DESTINATION,NETWORK_OUT_OF_ORDER,RECOVERY_ON_TIMER_EXPIRE"/>

       <action application="bridge" data="sofia/myprofile/$1 at x.x.x.x"/>

       <action application="bridge" data="sofia/myprofile/$1 at y.y.y.y"/>

       <action application="bridge" data="sofia/myprofile/$1 at z.z.z.z"/>

    </condition>

  </extension>

</include>

 

 

However if I want to use "|" between my gateways - and the example below is not working. And no matter what x.x.x.x returns - it
will try y.y.y.y  and eventually z.z.z.z.

 

<action application="set"

data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,SERVICE_NOT_IMPLEMENTED,CALL_RE

JECTED,NO_USER_RESPONSE,EXCHANGE_ROUTING_ERROR,INCOMPATIBLE_DESTINATION,NETWORK_OUT_OF_ORDER,RECOVERY_ON_TIMER_EXPIRE"/>

<action application="bridge"

data="sofia/myprofile/$1 at x.x.x.x|sofia/myprofile/$1 at y.y.y.y|sofia/myprofile/$1 at z.z.z.z"/>

 

Unfortunately, we have to use pipe for fail-over since we are using mod_lcr to choose between outbound gateways.

 

Any suggestions? 

 

Thanks,

 

Ivan 

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From msc at freeswitch.org  Tue Jun 14 05:38:02 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 13 Jun 2011 18:38:02 -0700
Subject: [Freeswitch-users] FreeSWITCH Cookbook Question: Pocketsphinx
Message-ID: <BANLkTinWyn8HqMgiObEksmaYfUf3wW2RVA@mail.gmail.com>

Hello all!

I am looking for examples of mod_pocketsphinx being used in a real-world
application. If you are using pocketsphinx ASR for something useful and can
share some configs I would appreciate it. I'm soliciting ideas on how to
make the pocketsphinx recipe in the cookbook be as truly useful as possible.
I don't personally use it for anything and I've only seen the pizza demo in
action, so I'm asking our intrepid community members for some thoughts.
Email me off list, please.

Thanks!

-Michael
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From lrobot.qq at gmail.com  Tue Jun 14 06:40:10 2011
From: lrobot.qq at gmail.com (Qingquan Luo)
Date: Tue, 14 Jun 2011 10:40:10 +0800
Subject: [Freeswitch-users] in bypass_media mode,
	2833 payload type modified by freeswitch!
Message-ID: <BANLkTikkQeXj_afrEGzneY9WnW-L7cBScg@mail.gmail.com>

Hi All,

  I use freeswitch to bridge one incoming call to other target phone number.
  I use bypass_media mode.
  So the rtp is not go through the freeswitch,
  By I notice that When LegB reply 183 or 200 SDP message. Freeswitch
change it 2833 telephone-event payload type. and forwarding the message to
caller. This make the caller use wrong dtmf payload type to send dtmf

  What wrong with that?  How can fix it?
  Any information or help is welcome.


Thanks


Best Regards


Qingquan



-- 
Using Gmail? Please read this important notice:
http://www.fsf.org/campaigns/jstrap/gmail?40922.
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From kris at kriskinc.com  Tue Jun 14 06:44:29 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Mon, 13 Jun 2011 22:44:29 -0400
Subject: [Freeswitch-users] in bypass_media mode,
 2833 payload type modified by freeswitch!
In-Reply-To: <BANLkTikkQeXj_afrEGzneY9WnW-L7cBScg@mail.gmail.com>
References: <BANLkTikkQeXj_afrEGzneY9WnW-L7cBScg@mail.gmail.com>
Message-ID: <BANLkTi=xcH12qx-h1NtDx6k-qdBb2dJEOg@mail.gmail.com>

Perhaps you didn't get my earlier reply:

1)  Turn on siptrace with "sofia global siptrace on" from fs_cli.
2)  Provide a full SIP trace and console debug on http://pastebin.freeswitch.org

On Mon, Jun 13, 2011 at 10:40 PM, Qingquan Luo <lrobot.qq at gmail.com> wrote:
> Hi All,
> ? I use freeswitch to bridge one incoming call to other target phone number.
> ? I use bypass_media mode.
> ? So the rtp is not go through the freeswitch,
> ? By I notice that When LegB reply 183 or 200 SDP message. Freeswitch
> change it 2833 telephone-event payload type. and forwarding the message to
> caller. This make the caller use wrong dtmf payload type to send dtmf
> ? What wrong with that? ?How can fix it?
> ? Any information or help is welcome.
>
> Thanks
>
> Best Regards
>
> Qingquan
>
>
> --
> Using Gmail? Please read this important notice:
> http://www.fsf.org/campaigns/jstrap/gmail?40922.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner


From freeswitch at aastral.net  Tue Jun 14 07:48:58 2011
From: freeswitch at aastral.net (Bill W.)
Date: Mon, 13 Jun 2011 23:48:58 -0400
Subject: [Freeswitch-users] start_dtmf_generate during early media?
In-Reply-To: <BANLkTikge0n3rxzPu8RY5ZNAbTVad3NUQg@mail.gmail.com>
References: <1QVra4-0008Vr-8x@mail.aastral.net>	<BANLkTim5idCYjZLmkhTbXV_dquupQZZd7w@mail.gmail.com>	<1QWBRK-0007Rl-Qf@mail.aastral.net>
	<BANLkTikge0n3rxzPu8RY5ZNAbTVad3NUQg@mail.gmail.com>
Message-ID: <1QWKci-0005DQ-Fw@mail.aastral.net>

Hey Micheal,

Okay, I believe I've tracked it down to this:  If I set

<action application="export"
data="nolocal:execute_on_media=start_dtmf_generate true"/>
<action application="set" data="bridge_early_media=true"/>

Then inband dtmf is NOT generated on the B-leg. If I remove the
bridge_early_media=true, then inband DTMF IS generated on the b-leg.

Is this a bug or a feature? ;-)

Thanks!
Bill



On 6/13/11 2:32 PM, Michael Collins wrote:
> Can you pb the debug output of this call? You might also record the call
> and listen to the audio, just to confirm what's happening.
> 
> -MC
> 


From sharad at coraltele.com  Tue Jun 14 08:54:33 2011
From: sharad at coraltele.com (sharad)
Date: Tue, 14 Jun 2011 10:24:33 +0530
Subject: [Freeswitch-users] Voicemail Message - How t know
	howmanyareread & how many are unread in Inbox
References: <mailman.10243.1307524956.24555.freeswitch-users@lists.freeswitch.org><AA1BE0FA88F3421784DBD1D91A84E515@sharad><BANLkTikshbtzM67daHBePp7TB5pH5FjEyA@mail.gmail.com><E826C94A3C6BE44198FAF4D66B5AB4D81BDA4D5BF1@VA3DIAXVS351.RED001.local><BANLkTikBYBK5Jugw6dtqUL88bkkic-bFAA@mail.gmail.com><26DE8A8116AB4F5BB31F0675DDBFD1BF@sharad><BANLkTiknapraGXceJXfcf9RHJXBRpbE0QQ@mail.gmail.com><DFC9276BB70D4FB99BD4FF88483B2590@sharad>
	<BANLkTi=jMGpHPJNdRLXVvMEP09GuBX5dJg@mail.gmail.com>
Message-ID: <FC094B6ECA5D46E89D9DDA07C858CA75@sharad>

Ok Thanks for all your efforts..

Best Regards
Sharad


----- Original Message ----- 
From: "Christian Benke" <benkokakao at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Friday, June 10, 2011 2:14 PM
Subject: Re: [Freeswitch-users] Voicemail Message - How t know 
howmanyareread & how many are unread in Inbox


>> Plz advice further..
>
> Sorry, no idea then.
> But i assume, if a user just listens to a message but doesn't archive
> it right away, he want's to listen to it again at a later time, so the
> message should remain "new" - therefore the behaviour you report is
> not wrong imho(not updating "read_epoch" is not beautiful though).
>
>
> 



From wstephen80 at gmail.com  Tue Jun 14 11:00:49 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 14 Jun 2011 09:00:49 +0200
Subject: [Freeswitch-users] unload pcmu and pcma
In-Reply-To: <1308007214.2014.1.camel@gustavo-laptop>
References: <1308007214.2014.1.camel@gustavo-laptop>
Message-ID: <BANLkTinPEW3ZS+SwugWZFCyoEWYvnXXgww@mail.gmail.com>

Hi, I use Sangoma transcoding card only to handle G711-G729 calls!

Probably, what the Sangoma people say is that the codec G711 is embedded in
Freeswitch and you cannot move it to Sangoma hardware.

This means that Sangoma card do the G729-L16 transcoding (high cpu
consuming) and Freeswitch do the L16-G711 transcoding (is only a mapping,
not algorithm = very low cpu consuming).

For this reason, for example, in all G711-G711 calls the Sangoma card is not
involved. For these calls, to have the G711 in Freeswitch is more efficient
then to have the G711 in Sangoma card!

Stephen


On Tue, Jun 14, 2011 at 1:20 AM, Gustavo Espeche <
gustavo.espeche at easyipcall.com> wrote:

> Hi we buy a sangoma d100 and we are having trouble because sangoma
> support inform to us that we can't do a trascoding g711-->g729 because
> we can unload the codec g711 from freeswitch.
> Some one know how we can fix this issue?
> Best Regards.
>
> --
>
> Gustavo Espeche
> EasyIpCall S.R.L.
> www.easyipcall.com
> Bv Mitre 517 24? E
> Cordoba - Argentina
> Te: +54 - 351 - 4280633
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Tue Jun 14 11:02:08 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 14 Jun 2011 08:02:08 +0100
Subject: [Freeswitch-users] bypass_media problem
In-Reply-To: <BANLkTinMGHmdvhrgWXV8OMJadzPSOFcawg@mail.gmail.com>
References: <BANLkTinMGHmdvhrgWXV8OMJadzPSOFcawg@mail.gmail.com>
Message-ID: <BANLkTinw2HzVR_tpA3nQjts3Up5z2FRi1w@mail.gmail.com>

session:execute("set","inbound-late-negotiation=true")
session:execute("set","inbound-bypass-media=false")

This will have no effect. Once the channel is created drop the inbound-
part, that is only used on the SIP profile.

-Steve



On 13 June 2011 23:59, David Villasmil <david.villasmil.work at gmail.com>wrote:

> Hello Guys,
>
> I have a set where i'm receiving a call from a gw and sending to another
> gw. Both un-natted. If I set in my (lua) script bypass_media=false, call
> goes just fine, if i set it to true, FS sends "Temporary unavailable" to the
> B-side and CANCELs the A-side...
>
> Codec is fine as you will see in the pastebin..
>
>
> here's the trace:
>
> http://pastebin.freeswitch.org/16480
>
> and here's the console log:
>
> http://pastebin.freeswitch.org/16481
>
> I'm bridging with a lua script, this is the pertinent part:
>
> gw_sip_username and gw_sip_pwd are just bogus "1234","1234", no
> registration is needed....
>
>
>
>
>
>  session:execute("set","continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,407")
>
>      session:execute("set","inbound-late-negotiation=true")
>      session:execute("set","inbound-bypass-media=false")
>
>      session:execute("set","proxy_media=false")
>      session:execute("set","bypass_media=false")
>
>
>      --session:execute("bridge","{sip_auth_username=" .. gw_sip_username ..
> ",sip_auth_password=" .. gw_sip_pwd .. "}sofia/external/".. out_number
> .."@".. gw_sip_ip .."")
>
>
>      fsLog("BRIDGE EXECUTE:", "{loop=3}sofia/gateway/${distributor(" ..
> route_name .. ")}" .. out_number .. "")
>      session:execute("bridge","{loop=3}sofia/gateway/${distributor(" ..
> route_name .. ")}".. out_number .."")
>
>      -- hangup
>      session:hangup();
>
>
>
> Thanks guys for your help!
>
>
> David
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From maciej.aniserowicz at gmail.com  Tue Jun 14 11:40:00 2011
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Tue, 14 Jun 2011 00:40:00 -0700 (PDT)
Subject: [Freeswitch-users] Eavesdrop between 2 FS instaces
In-Reply-To: <BANLkTi=aMvSa9Ey=hzXzoAsJEGNqZcpbCg@mail.gmail.com>
References: <1307039305975-6432366.post@n2.nabble.com>
	<BANLkTinLdaHeaMy9XCthLM87jxQKNGSBXw@mail.gmail.com>
	<1307106103579-6435172.post@n2.nabble.com>
	<BANLkTi=aMvSa9Ey=hzXzoAsJEGNqZcpbCg@mail.gmail.com>
Message-ID: <1308037200643-6473071.post@n2.nabble.com>

Yes, I tried this also...
3 applications are needed here:

(command sent to FS1)
[app1] [CallA channel] [app2]:{sip_h_X-target_channel_id=[CallB
 channel]}sofia/bridging/123456 at ip:port,park inline

(dialplan in FS2)
<condition field="destination_number" expression="^(123456)$">
    &lt;action application=&quot;&lt;b&gt;app3"
data="${sip_h_X-target_channel_id}" />
 </condition>

app1 is uuid_transfer
app3 is eavesdrop
app2...? I cannot find anything that fits in here.

MA



Steven Ayre wrote:
> 
> I think you want the eavesdrop app:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop
> 
> Intercept takes over the call. Eavesdrop just silently listens in.
> 
> -Steve
> 
> 
> 
> On 3 June 2011 14:01, Maciej Aniserowicz
> &lt;maciej.aniserowicz at gmail.com&gt;wrote:
> 
>> Hi Steve,
>> Thanks for interest.
>>
>> Yes, i figured out that i cannot eavesdrop any 'easy' way. I tried the
>> approach you described (dialing into FS2 to a special context and passing
>> sip-h_X header). In fact i use this to bridge calls on different FSes.
>> But
>> while i managed to get it working with bridging, i cannot find a way to
>> do
>> eavesdropping.
>>
>> With bridging I send:
>> uuid_transfer [CallA channel] bridge:{sip_h_X-target_channel_id=[CallB
>> channel]}sofia/bridging/123456 at ip:port,park inline
>>
>> Then FS2 in dialplan for 123456 extracts target channel id from sip
>> header
>> and intercepts it:
>> <condition field="destination_number" expression="^(123456)$">
>>    <action application="intercept" data="${sip_h_X-target_channel_id}" />
>> </condition>
>>
>> What actions should i define in FS2 dialplan for eavesdropping? What
>> inline
>> dialplan should i send in dial string? I'm stuck here.
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/Eavesdrop-between-2-FS-instaces-tp6432366p6435172.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Eavesdrop-between-2-FS-instaces-tp6432366p6473071.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From valery.kalinin at gmail.com  Tue Jun 14 11:57:35 2011
From: valery.kalinin at gmail.com (Valery Kalinin)
Date: Tue, 14 Jun 2011 13:57:35 +0600
Subject: [Freeswitch-users] Cannot configure ftmod_libpri
Message-ID: <BANLkTi=rKME7_36cYUoodVObNt5ZNq=gew@mail.gmail.com>

Hi all!

Cannot setup FreeTDM with libpri.
I have configured mode network on channel E1, the provider also set this
mode.
But log write:
2011-06-14 13:42:01.930818 [WARNING] ftmod_libpri.c:1590 -- PRI error event:
We think we're the network, but they think they're the network, too.
2011-06-14 13:42:02.934635 [WARNING] ftmod_libpri.c:1590 -- PRI error event:
We think we're the network, but they think they're the network, too.
2011-06-14 13:42:03.965449 [WARNING] ftmod_libpri.c:1590 -- PRI error event:
We think we're the network, but they think they're the network, too.
2011-06-14 13:42:04.965269 [WARNING] ftmod_libpri.c:1590 -- PRI error event:
We think we're the network, but they think they're the network, too.
2011-06-14 13:42:05.905336 [WARNING] ftmod_libpri.c:1590 -- PRI error event:
We think we're the network, but they think they're the network, too.
2011-06-14 13:42:06.925361 [WARNING] ftmod_libpri.c:1590 -- PRI error event:
We think we're the network, but they think they're the network, too.
2011-06-14 13:42:07.945387 [WARNING] ftmod_libpri.c:1590 -- PRI error event:
We think we're the network, but they think they're the network, too.
2011-06-14 13:42:08.565403 [CRIT] lpwrap_pri.c:321 Error = -1 [Invalid or
incomplete multibyte or wide character]
2011-06-14 13:43:07.975737 [ERR] Span:0 Received UA frame in invalid state

   <libpri_spans>
     <span id="1" name="pri">
    <param name="node" value="network"/>
    <param name="switch" value="4ess"/>
    <param name="opts" value="omit_redirecting_number"/>
    <param name="l1" value="alaw"/>
    <param name="dp" value="unknown"/>
    <param name="debug" value="all"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="default"/>
     </span>
   </libpri_spans>

How to properly set up FreeTDM?
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From steveayre at gmail.com  Tue Jun 14 12:14:08 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 14 Jun 2011 09:14:08 +0100
Subject: [Freeswitch-users] bypass_media problem
In-Reply-To: <BANLkTinMGHmdvhrgWXV8OMJadzPSOFcawg@mail.gmail.com>
References: <BANLkTinMGHmdvhrgWXV8OMJadzPSOFcawg@mail.gmail.com>
Message-ID: <BANLkTin8EJuhEgKNov5urDJD-F33BVCQSg@mail.gmail.com>

Can't see the reason immediately... the channel appears to be hanging up
because it thinks the bridge is finished.

Can you repeat the debug log, but this time enabling FreeSWITCH's siptrace
option? (sofia global siptrace on).

Also, I this isn't doing what I think you expect:
session:execute("bridge","{loop=3}sofia/gateway/${distributor(" ..
route_name .. ")}".. out_number .."")
The distributor API is called just once to generate the string to give to
the bridge app, not on each loop. So you'll be dialing through the same
gateway 3 times not rerouting to another gateway.

Also, what FS version is this?

-Steve


On 13 June 2011 23:59, David Villasmil <david.villasmil.work at gmail.com>wrote:

> Hello Guys,
>
> I have a set where i'm receiving a call from a gw and sending to another
> gw. Both un-natted. If I set in my (lua) script bypass_media=false, call
> goes just fine, if i set it to true, FS sends "Temporary unavailable" to the
> B-side and CANCELs the A-side...
>
> Codec is fine as you will see in the pastebin..
>
> rou
> here's the trace:
>
> http://pastebin.freeswitch.org/16480
>
> and here's the console log:
>
> http://pastebin.freeswitch.org/16481
>
> I'm bridging with a lua script, this is the pertinent part:
>
> gw_sip_username and gw_sip_pwd are just bogus "1234","1234", no
> registration is needed....
>
>
>
>
>
>  session:execute("set","continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,407")
>
>      session:execute("set","inbound-late-negotiation=true")
>      session:execute("set","inbound-bypass-media=false")
>
>      session:execute("set","proxy_media=false")
>      session:execute("set","bypass_media=false")
>
>
>      --session:execute("bridge","{sip_auth_username=" .. gw_sip_username ..
> ",sip_auth_password=" .. gw_sip_pwd .. "}sofia/external/".. out_number
> .."@".. gw_sip_ip .."")
>
>
>      fsLog("BRIDGE EXECUTE:", "{loop=3}sofia/gateway/${distributor(" ..
> route_name .. ")}" .. out_number .. "")
>      session:execute("bridge","{loop=3}sofia/gateway/${distributor(" ..
> route_name .. ")}".. out_number .."")
>
>      -- hangup
>      session:hangup();
>
>
>
> Thanks guys for your help!
>
>
> David
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Tue Jun 14 12:21:50 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 14 Jun 2011 09:21:50 +0100
Subject: [Freeswitch-users] Cannot configure ftmod_libpri
In-Reply-To: <BANLkTi=rKME7_36cYUoodVObNt5ZNq=gew@mail.gmail.com>
References: <BANLkTi=rKME7_36cYUoodVObNt5ZNq=gew@mail.gmail.com>
Message-ID: <BANLkTinNLZ2+M0JQZTWSZMJmSGjk6fO5ag@mail.gmail.com>

    <param name="node" value="network"/>

Try changing that to:

    <param name="node" value="cpe"/>

One side has to be network, the other has to be cpe (like master-slave).
That error makes it look like you're both trying to be the network end.

-Steve


On 14 June 2011 08:57, Valery Kalinin <valery.kalinin at gmail.com> wrote:

> Hi all!
>
> Cannot setup FreeTDM with libpri.
> I have configured mode network on channel E1, the provider also set this
> mode.
> But log write:
> 2011-06-14 13:42:01.930818 [WARNING] ftmod_libpri.c:1590 -- PRI error
> event: We think we're the network, but they think they're the network, too.
> 2011-06-14 13:42:02.934635 [WARNING] ftmod_libpri.c:1590 -- PRI error
> event: We think we're the network, but they think they're the network, too.
> 2011-06-14 13:42:03.965449 [WARNING] ftmod_libpri.c:1590 -- PRI error
> event: We think we're the network, but they think they're the network, too.
> 2011-06-14 13:42:04.965269 [WARNING] ftmod_libpri.c:1590 -- PRI error
> event: We think we're the network, but they think they're the network, too.
> 2011-06-14 13:42:05.905336 [WARNING] ftmod_libpri.c:1590 -- PRI error
> event: We think we're the network, but they think they're the network, too.
> 2011-06-14 13:42:06.925361 [WARNING] ftmod_libpri.c:1590 -- PRI error
> event: We think we're the network, but they think they're the network, too.
> 2011-06-14 13:42:07.945387 [WARNING] ftmod_libpri.c:1590 -- PRI error
> event: We think we're the network, but they think they're the network, too.
> 2011-06-14 13:42:08.565403 [CRIT] lpwrap_pri.c:321 Error = -1 [Invalid or
> incomplete multibyte or wide character]
> 2011-06-14 13:43:07.975737 [ERR] Span:0 Received UA frame in invalid state
>
>    <libpri_spans>
>      <span id="1" name="pri">
>     <param name="node" value="network"/>
>     <param name="switch" value="4ess"/>
>     <param name="opts" value="omit_redirecting_number"/>
>     <param name="l1" value="alaw"/>
>     <param name="dp" value="unknown"/>
>     <param name="debug" value="all"/>
>     <param name="dialplan" value="XML"/>
>     <param name="context" value="default"/>
>      </span>
>    </libpri_spans>
>
> How to properly set up FreeTDM?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ankitwalia4u at gmail.com  Tue Jun 14 12:38:28 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Tue, 14 Jun 2011 14:08:28 +0530
Subject: [Freeswitch-users] Adding a New User
Message-ID: <BANLkTi=QJ7sAgP3dj6Zj-R7zQoLYZaoKdA@mail.gmail.com>

Dear all,

I am new to FreeSwitch. I added a new user ext 1100 by below two steps.

1. Create a new XML file for the user, usually by copying an existing file
2. Modify the Local_Extension Dialplan entry

Now, I tried to connect to the FS Server using SIP protocol and Soft Phone
X-lite. On X-lite, it is giving error "Account cant be enable"

At FS CLI, it is giving the error
freeswitch at ankit-PC> 2011-06-14 13:37:45.695430 [WARNING] sofia_reg.c:1872
Can't
 find user [1100 at 192.168.1.5]
You must define a domain called '192.168.1.5' in your directory and add a
user w
ith the id="1100" attribute
and you must configure your device to use the proper domain in it's
authenticati
on credentials.
2011-06-14 13:37:45.703430 [WARNING] sofia_reg.c:1030 SIP auth failure
(REGISTER
) on sofia profile 'internal' for [1100 at 192.168.1.5] from ip 192.168.1.5

I reloaded the XML also.

My SIP soft phone and my Freeswitch server is at the same machine having
same IP address.

I did some hit and trial and also tried to find on mailing list. I could not
find what could be the problem.

Please help.

Thanks
Ankit

-- 
Life is like a rose its upto u feel it as its fragrance or thorns
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From steveayre at gmail.com  Tue Jun 14 12:41:53 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 14 Jun 2011 09:41:53 +0100
Subject: [Freeswitch-users] Eavesdrop between 2 FS instaces
In-Reply-To: <1308037200643-6473071.post@n2.nabble.com>
References: <1307039305975-6432366.post@n2.nabble.com>
	<BANLkTinLdaHeaMy9XCthLM87jxQKNGSBXw@mail.gmail.com>
	<1307106103579-6435172.post@n2.nabble.com>
	<BANLkTi=aMvSa9Ey=hzXzoAsJEGNqZcpbCg@mail.gmail.com>
	<1308037200643-6473071.post@n2.nabble.com>
Message-ID: <BANLkTinKXKfao5i=g9OkQMc9dOehA2cfkw@mail.gmail.com>

Try something like this:

FS1:
uuid_broadcast <uuid>
bridge:{sip_h_X-target_
channel_id=<uuid>}sofia/bridging/123456 at ip:port aleg

FS2:
<condition field="destination_number" expression="^(123456)$">
  <action application="eavesdrop" data="${sip_h_X-target_channel_id}" />
</condition>

-Steve


On 14 June 2011 08:40, Maciej Aniserowicz <maciej.aniserowicz at gmail.com>wrote:

> Yes, I tried this also...
> 3 applications are needed here:
>
> (command sent to FS1)
> [app1] [CallA channel] [app2]:{sip_h_X-target_channel_id=[CallB
>  channel]}sofia/bridging/123456 at ip:port,park inline
>
> (dialplan in FS2)
> <condition field="destination_number" expression="^(123456)$">
>     &lt;action application=&quot;&lt;b&gt;app3"
> data="${sip_h_X-target_channel_id}" />
>  </condition>
>
> app1 is uuid_transfer
> app3 is eavesdrop
> app2...? I cannot find anything that fits in here.
>
> MA
>
>
>
> Steven Ayre wrote:
> >
> > I think you want the eavesdrop app:
> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop
> >
> > Intercept takes over the call. Eavesdrop just silently listens in.
> >
> > -Steve
> >
> >
> >
> > On 3 June 2011 14:01, Maciej Aniserowicz
> > &lt;maciej.aniserowicz at gmail.com&gt;wrote:
> >
> >> Hi Steve,
> >> Thanks for interest.
> >>
> >> Yes, i figured out that i cannot eavesdrop any 'easy' way. I tried the
> >> approach you described (dialing into FS2 to a special context and
> passing
> >> sip-h_X header). In fact i use this to bridge calls on different FSes.
> >> But
> >> while i managed to get it working with bridging, i cannot find a way to
> >> do
> >> eavesdropping.
> >>
> >> With bridging I send:
> >> uuid_transfer [CallA channel] bridge:{sip_h_X-target_channel_id=[CallB
> >> channel]}sofia/bridging/123456 at ip:port,park inline
> >>
> >> Then FS2 in dialplan for 123456 extracts target channel id from sip
> >> header
> >> and intercepts it:
> >> <condition field="destination_number" expression="^(123456)$">
> >>    <action application="intercept" data="${sip_h_X-target_channel_id}"
> />
> >> </condition>
> >>
> >> What actions should i define in FS2 dialplan for eavesdropping? What
> >> inline
> >> dialplan should i send in dial string? I'm stuck here.
> >>
> >> --
> >> View this message in context:
> >>
> http://freeswitch-users.2379917.n2.nabble.com/Eavesdrop-between-2-FS-instaces-tp6432366p6435172.html
> >> Sent from the freeswitch-users mailing list archive at Nabble.com.
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Eavesdrop-between-2-FS-instaces-tp6432366p6473071.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From valery.kalinin at gmail.com  Tue Jun 14 15:00:33 2011
From: valery.kalinin at gmail.com (Valery Kalinin)
Date: Tue, 14 Jun 2011 17:00:33 +0600
Subject: [Freeswitch-users]  Cannot configure ftmod_libpri
Message-ID: <BANLkTim1opMjOLOEM_Ebtt5e5Ge22ZEz0A@mail.gmail.com>

Now other errors:

2011-06-14 16:51:31.958295 [WARNING] ftdm_io.c:2753 [s1c19][1:20] Channel
not opened, proceeding anyway
2011-06-14 16:51:31.958295 [WARNING] ftdm_io.c:2753 [s1c20][1:21] Channel
not opened, proceeding anyway
2011-06-14 16:51:31.958295 [WARNING] ftdm_io.c:2753 [s1c21][1:22] Channel
not opened, proceeding anyway
2011-06-14 16:51:31.958295 [WARNING] ftdm_io.c:2753 [s1c22][1:23] Channel
not opened, proceeding anyway
2011-06-14 16:51:31.958295 [WARNING] ftdm_io.c:2753 [s1c23][1:24] Channel
not opened, proceeding anyway
2011-06-14 16:51:31.958295 [WARNING] ftdm_io.c:2753 [s1c24][1:25] Channel
not opened, proceeding anyway
2011-06-14 16:51:31.958295 [WARNING] ftdm_io.c:2753 [s1c25][1:26] Channel
not opened, proceeding anyway
2011-06-14 16:51:31.958295 [WARNING] ftdm_io.c:2753 [s1c26][1:27] Channel
not opened, proceeding anyway
2011-06-14 16:51:31.958295 [WARNING] ftdm_io.c:2753 [s1c27][1:28] Channel
not opened, proceeding anyway
2011-06-14 16:51:31.958295 [WARNING] ftdm_io.c:2753 [s1c28][1:29] Channel
not opened, proceeding anyway
2011-06-14 16:51:31.958295 [WARNING] ftdm_io.c:2753 [s1c29][1:30] Channel
not opened, proceeding anyway
2011-06-14 16:51:31.958295 [WARNING] ftdm_io.c:2753 [s1c30][1:31] Channel
not opened, proceeding anyway
2011-06-14 16:51:32.058312 [ERR] ftmod_zt.c:1075 [s1c31][1:16] Bad frame
checksum (ZT_EVENT_BADFCS)
2011-06-14 16:51:33.588339 [ERR] ftmod_zt.c:1087 [s1c31][1:16] HDLC abort
frame received (ZT_EVENT_ABORT)
2011-06-14 16:51:33.598348 [ERR] ftmod_zt.c:1075 [s1c31][1:16] Bad frame
checksum (ZT_EVENT_BADFCS)
2011-06-14 16:51:33.608355 [ERR] ftmod_zt.c:1075 [s1c31][1:16] Bad frame
checksum (ZT_EVENT_BADFCS)
2011-06-14 16:51:36.158405 [ERR] ftmod_zt.c:1075 [s1c31][1:16] Bad frame
checksum (ZT_EVENT_BADFCS)
2011-06-14 16:51:36.178402 [ERR] ftmod_zt.c:1075 [s1c31][1:16] Bad frame
checksum (ZT_EVENT_BADFCS)
2011-06-14 16:51:36.188409 [ERR] ftmod_zt.c:1075 [s1c31][1:16] Bad frame
checksum (ZT_EVENT_BADFCS)
2011-06-14 16:51:36.208410 [ERR] ftmod_zt.c:1075 [s1c31][1:16] Bad frame
checksum (ZT_EVENT_BADFCS)
2011-06-14 16:51:36.258408 [ERR] ftmod_zt.c:1075 [s1c31][1:16] Bad frame
checksum (ZT_EVENT_BADFCS)
2011-06-14 16:51:36.358406 [ERR] ftmod_zt.c:1075 [s1c31][1:16] Bad frame
checksum (ZT_EVENT_BADFCS)
2011-06-14 16:51:36.388410 [ERR] ftmod_zt.c:1087 [s1c31][1:16] HDLC abort
frame received (ZT_EVENT_ABORT)
2011-06-14 16:51:36.438412 [ERR] ftmod_zt.c:1075 [s1c31][1:16] Bad frame
checksum (ZT_EVENT_BADFCS)
2011-06-14 16:51:36.498413 [ERR] ftmod_zt.c:1075 [s1c31][1:16] Bad frame
checksum (ZT_EVENT_BADFCS)
2011-06-14 16:51:36.868433 [WARNING] sofia_reg.c:1241 SIP auth challenge
(INVITE) on sofia profile 'internal' for [2002 at 192.168.205.1] from ip
192.168.20
2011-06-14 16:51:36.928427 [WARNING] sofia_glue.c:213 Codec G723 payload 4
added to sdp wanting ptime 30 but it's already 20 (PCMA:8:20), disabling
ptime
2011-06-14 16:51:41.031529 [ERR] ftmod_zt.c:1075 [s1c31][1:16] Bad frame
checksum (ZT_EVENT_BADFCS)
2011-06-14 16:51:43.078575 [ERR] ftmod_zt.c:1075 [s1c31][1:16] Bad frame
checksum (ZT_EVENT_BADFCS)
2011-06-14 16:51:44.961433 [CRIT] lpwrap_pri.c:321 Error = -1 [Invalid or
incomplete multibyte or wide character]
2011-06-14 16:52:12.490765 [CRIT] sofia.c:841 No more sessions allowed at
this time.
2011-06-14 16:52:12.649384 [ERR] Span:0 Received UA frame in invalid state
2011-06-14 16:52:15.057934 [WARNING] ozmod_zt.c:1072 Unhandled event 8 for
1:16
2011-06-14 16:52:15.157916 [WARNING] ozmod_zt.c:1072 Unhandled event 8 for
1:16
2011-06-14 16:52:15.257898 [WARNING] ozmod_zt.c:1072 Unhandled event 8 for
1:16
2011-06-14 16:52:15.357880 [WARNING] ozmod_zt.c:1072 Unhandled event 8 for
1:16
2011-06-14 16:52:15.457862 [WARNING] ozmod_zt.c:1072 Unhandled event 6 for
1:16
2011-06-14 16:52:15.557844 [WARNING] ozmod_zt.c:1072 Unhandled event 6 for
1:16
2011-06-14 16:52:15.657826 [WARNING] ozmod_zt.c:1072 Unhandled event 6 for
1:16
2011-06-14 16:52:15.757808 [WARNING] ozmod_zt.c:1072 Unhandled event 6 for
1:16
2011-06-14 16:52:15.857790 [WARNING] ozmod_zt.c:1072 Unhandled event 6 for
1:16
2011-06-14 16:52:16.057754 [WARNING] ozmod_zt.c:1072 Unhandled event 6 for
1:16
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From ankitwalia4u at gmail.com  Tue Jun 14 15:51:28 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Tue, 14 Jun 2011 17:21:28 +0530
Subject: [Freeswitch-users] Invalid Profile [external]
Message-ID: <BANLkTinMcV_hY0yrg-FeD+4Aag+u+oB_=w@mail.gmail.com>

Dear all,

I am new to FreeSwitch. I am trying to use an external SIP profile for
connecting ouside my LAN

I create a SIP account on iptel.org.

To add a new gateway, I created a file and saved into the External folder

<include>
<gateway name="custom">
<param name="username" value="MY_USER_NAME"/>
<param name="password" value="MY_PASSWORD"/>
<param name="realm" value="iptel.org"/>
<!-- iptel.org requires a 'proxy' parameter -->
<param name="proxy" value="sip.iptel.org"/>
</gateway>
</include>

I have given the username and password which I got after creating an account
at iptel.org.

Now, after launching FS CLI, I run a "sofia profile external restart
reloadxml" which is giving me message "Invalid Profile [external]"

I am not able to make up what is the problem.

I have some understanding of Networking. When i tried to find this issue on
internet I found that this could be related to:
1. Is this a problem because of NAT? or Port Forwarding which I need to do?

Please give me some clue to debug the issue.

Thanks
Ankit



-- 
Life is like a rose its upto u feel it as its fragrance or thorns
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From gustavo.espeche at easyipcall.com  Tue Jun 14 16:35:26 2011
From: gustavo.espeche at easyipcall.com (Gustavo Espeche)
Date: Tue, 14 Jun 2011 09:35:26 -0300
Subject: [Freeswitch-users] Freeswitch transcoding
Message-ID: <1308054926.3665.6.camel@gustavo-laptop>

Hi we are trying to do transcoding with freeswitch g711a-->g711u or
g711-->speex but we can't do that FS do a trascoding, follow is a
dialplan that we use with mod_curl

<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<section name="dialplan" description="Regex/XML Dialplan">
<context name="public">
<extension name="external">
<condition field="destination_number" expression="^666654351428063">
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="progress_timeout=20"/>
<action application="set"
data="export_vars=et_routing_number,et_routing_host,et_routed_number,et_routed_host,et_random_key"/>
<action application="set" data="et_routing_number=666654351428063"/>
<action application="set" data="et_routing_host=10.0.0.12"/>
<action application="set" data="et_random_key=029645322327599477"/>
<action application="set" data="proxy_media=false"/>
<action application="set" data="bypass_media=true"/>
<action application="set" data="absolute_codec_string=PCMU"/>
<action application="export" data="nolocal:absolute_codec_string=PCMA"/>
<action application="set" data="et_routed_number=555554351428063"/>
<action application="set" data="et_routed_host=10.0.0.14"/>
<action application="bridge"
data="sofia/external/555554351428063 at 10.0.0.14"/>
</condition>
</extension>
</context>
</section>
</document>

 we relay can't understand what are we doing wrong.
are we need configure someone special in FS? we are using
sip_external_profile with port 5060.


Regards.

-- 

Gustavo Espeche
EasyIpCall S.R.L.
www.easyipcall.com
Bv Mitre 517 24? E
Cordoba - Argentina
Te: +54 - 351 - 4280633



From benkokakao at gmail.com  Tue Jun 14 16:42:32 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Tue, 14 Jun 2011 14:42:32 +0200
Subject: [Freeswitch-users] Adding a New User
In-Reply-To: <BANLkTi=QJ7sAgP3dj6Zj-R7zQoLYZaoKdA@mail.gmail.com>
References: <BANLkTi=QJ7sAgP3dj6Zj-R7zQoLYZaoKdA@mail.gmail.com>
Message-ID: <BANLkTinhiMjEDaM_okCP1f3H1VifvEmOfA@mail.gmail.com>

> freeswitch at ankit-PC> 2011-06-14 13:37:45.695430 [WARNING] sofia_reg.c:1872
> Can't
> ?find user [1100 at 192.168.1.5]
> You must define a domain called '192.168.1.5' in your directory and add a
> user w
> ith the id="1100" attribute
> and you must configure your device to use the proper domain in it's
> authenticati
> on credentials.
> 2011-06-14 13:37:45.703430 [WARNING] sofia_reg.c:1030 SIP auth failure
> (REGISTER
> ) on sofia profile 'internal' for [1100 at 192.168.1.5] from ip 192.168.1.5
> I reloaded the XML also.

Hi Ankit!

Ignore the X-Lite message - this is a (minor) configuration error in FreeSWITCH.

Is there more than one network-device configured on the machine?

If yes, open /usr/local/freeswitch/conf/vars.xml and verify that you
have the following settings somewhere:

<!--IP Adress-->
<X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.1.5"/>

<!-- Domain -->
<!-- Sets the default domain. -->
<X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
<X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/>

Best regards
Christian


From benkokakao at gmail.com  Tue Jun 14 16:51:09 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Tue, 14 Jun 2011 14:51:09 +0200
Subject: [Freeswitch-users] Invalid Profile [external]
In-Reply-To: <BANLkTinMcV_hY0yrg-FeD+4Aag+u+oB_=w@mail.gmail.com>
References: <BANLkTinMcV_hY0yrg-FeD+4Aag+u+oB_=w@mail.gmail.com>
Message-ID: <BANLkTimqAa=4+m6ni6Exac5xX9fU8eSiKA@mail.gmail.com>

> Now, after launching FS CLI, I run a "sofia profile external restart
> reloadxml" which is giving me message "Invalid Profile [external]"

Please give me the full output of the command above, i believe it says
more than just "Invalid Profile".

Best regards,
Christian


From educs13 at yahoo.com.br  Tue Jun 14 16:59:59 2011
From: educs13 at yahoo.com.br (educs13)
Date: Tue, 14 Jun 2011 05:59:59 -0700 (PDT)
Subject: [Freeswitch-users] Lua ASR TTS
In-Reply-To: <BANLkTimUAy28CHKTA4rm_31bA5h+_PBVJw@mail.gmail.com>
References: <1307202350274-6439449.post@n2.nabble.com>
	<BANLkTimUAy28CHKTA4rm_31bA5h+_PBVJw@mail.gmail.com>
Message-ID: <597923.84033.qm@web161919.mail.bf1.yahoo.com>

Hi MC,

thank you very much! I tried all the following combinations:
i) break;
ii) break all;
iii) uuid_break;
iv) uuid_break all;
but I always had the same problem...

As a "workaround", I changed my script this way:

1) I play the prompt:
session:streamFile("/usr/local/freeswitch/sounds/en/us/pizza/GP-DeliveryorTakeout.wav");
2) I play a short sample of silence. 
session:streamFile("/home/ubuntu/audio/silence.wav");??? 

With this strategy, if I say something when the prompt is being played, it is stopped and a short sample (duration: ~20 ms) without audio is played. It's not the ideal solution, but, at least, it doesn't stuck.

If you have another idea, please tell me. I would like to solve it without "workarounds".

Eduardo



--- Em seg, 13/6/11, mercutioviz [via freeswitch-users] <ml-node+6471843-2137856909-339117 at n2.nabble.com> escreveu:

De: mercutioviz [via freeswitch-users] <ml-node+6471843-2137856909-339117 at n2.nabble.com>
Assunto: Re: Lua ASR TTS
Para: "educs13" <educs13 at yahoo.com.br>
Data: Segunda-feira, 13 de Junho de 2011, 18:14



	Sorry for the late reply - I missed this one. Are you using the break app or the uuid_break API? Also, are you using the "all" argument?-MC

On Sat, Jun 4, 2011 at 8:45 AM, educs13 <[hidden email]> wrote:

Hi,



I was trying to run the 'Examples directory lua asr tts'

(http://wiki.freeswitch.org/wiki/Examples_directory_lua_asr_tts) in Ubuntu

10.04/32 bits, but I had some problems with the 'session:streamFile()'.



If I say something when the prompt is still playing, the audio stucks

indefinitely. It seems that the 'break' command isn't doing all its job. I

guess that maybe the 'break' is just stop feeding the audio player with new

samples, but it is not stopping the player. So, this player keeps reading

and playing the same samples continuously ...



I really don't know if my guess makes sense and as I don't know C very well,

I don't know where to look for a solution in the source code. I Hope you

could help me.



Thanks,

Eduardo







--

View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Lua-ASR-TTS-tp6439449p6439449.html


Sent from the freeswitch-users mailing list archive at Nabble.com.



_______________________________________________

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		If you reply to this email, your message will be added to the discussion below:
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From dujinfang at gmail.com  Tue Jun 14 17:09:53 2011
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 14 Jun 2011 21:09:53 +0800
Subject: [Freeswitch-users] interesting mod_erlang_event performance problem
Message-ID: <B8DE0D286CF341DAA1E93043BFF0453F@gmail.com>

Hi,

First, i'm running on Mac 10.6.7 with FreeSWITCH Version 1.0.head (git-765908f 2011-05-22 19-10-52 -0500)

I found erlang receives events really slow so I did some tests.

I use two custom erl to receive events, the file is on github:

https://gist.github.com/1024808

The only difference of fse.erl and fse2.erl is that the following line in fse.erl was commented (line 16).

{blah, 'freeswitch at localhost'} ! {blah}, % comment this line will be slow

I run the scripts at about the same time, so they should can get the same events. The problem is that fse2.erl runs fast ( 1000 events in 6s ) while fse.erl is slow (1000 events in 2 min).


(s2 at localhost)1> fse2:start().
start {{2011,6,14},{20,21,41}}
end {{2011,6,14},{20,21,47}}
ok

(s at localhost)1> fse:start().
start {{2011,6,14},{20,21,36}}
end {{2011,6,14},{20,23,24}}
ok

It's like fse.erl can only get 10 events per sec, so I guess there's some sleep(100) things in code, but sending a message(what ever) to FS breaks the sleep.

the {blah} message can be anything, while a valid msg can get ok, an invalid one gets {error, undef} etc.


I use the simple shell script to generate many events:

#!/bin/bash
IP=192.168.7.7

for f in `seq 1 8`; do
for f in `seq 1 20`; do
fs_cli -x "bgapi originate sofia/internal/load_test@$IP:5080 9664"
done
sleep 2
done



-- 
Seven Du
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj: http://www.freeswitch.org.cn
Sent with Sparrow
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From benkokakao at gmail.com  Tue Jun 14 17:10:16 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Tue, 14 Jun 2011 15:10:16 +0200
Subject: [Freeswitch-users] Invalid Profile [external]
In-Reply-To: <BANLkTimqAa=4+m6ni6Exac5xX9fU8eSiKA@mail.gmail.com>
References: <BANLkTinMcV_hY0yrg-FeD+4Aag+u+oB_=w@mail.gmail.com>
	<BANLkTimqAa=4+m6ni6Exac5xX9fU8eSiKA@mail.gmail.com>
Message-ID: <BANLkTi=BxWtAbiBGnfAtu+x2U2qhiGZDbQ@mail.gmail.com>

On 14 June 2011 14:51, Christian Benke <benkokakao at gmail.com> wrote:
>> Now, after launching FS CLI, I run a "sofia profile external restart
>> reloadxml" which is giving me message "Invalid Profile [external]"
>
> Please give me the full output of the command above, i believe it says
> more than just "Invalid Profile".

Oh! It's indeed just one line saying "2011-06-14 13:07:47.335914
[WARNING] sofia.c:4044 No Such Profile 'external'"

You probably have a typo in your sip-profile-configuration, in case
you didn't change it intentionally.

The default configuration for the external sip-profile is
/usr/local/freeswitch/conf/sip_profiles/external.xml(Modify the path
according to your Win7-installation).

Please verify you have a section in your configuration that says:

<profile name="external">
  <gateways>
    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
  </gateways>

...lots of other stuff in default-config...

</profile>


Then you add the iptel-configuration you've posted above in
/usr/local/freeswitch/conf/sip_profiles/external/iptel.xml(If this
didn't happen already)

Best regards,
Christian


From steveayre at gmail.com  Tue Jun 14 17:14:08 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 14 Jun 2011 14:14:08 +0100
Subject: [Freeswitch-users] Freeswitch transcoding
In-Reply-To: <1308054926.3665.6.camel@gustavo-laptop>
References: <1308054926.3665.6.camel@gustavo-laptop>
Message-ID: <BANLkTik1MXW6DcNYk3GbExR=w_6n623wTg@mail.gmail.com>

At a guess, it's this line:
<action application="set" data="bypass_media=true"/>
Unless the media is going through FS it can't trancode. The SDP (codec list)
will be forwarded from the aleg to the bleg so absolute_codec_string would
be ignored.

Also this has no effect, unless have proxy_media set to true on the sip
profile. You don't need to set them to false, they'll already be false.
You're correct to have this disabled though, as that would also prevent
transcoding.
<action application="set" data="proxy_media=false"/>

This won't do as you think:
<action application="set" data="absolute_codec_string=PCMU"/>
It chooses the codec before you get to the dialplan. absolute_codec_string
will only apply to the bleg. The incoming codec will either be picked from
the inbound-codec-prefs param on the sip profile that receives the a-leg, or
if you're using late-negotiation it'll pick that after the bridge so that
it'll try to use the same codec as the bleg if possible and pick one of the
other choices and transcode if that's not possible. absolute_codec_string
overrides the outbound-codec-prefs sip profile param on the bleg.

Try these on your sip profile:
 <param name="inbound-codec-prefs" value="PCMU" />
 <param name="outbound-codec-prefs" value="PCMA" />

Then this will transcode:
<extension name="external">
     <condition field="destination_number" expression="^666654351428063">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="progress_timeout=20"/>
          <action application="set"
data="export_vars=et_routing_number,et_routing_host,et_routed_number,et_routed_host,et_random_key"/>
          <action application="set"
data="et_routing_number=666654351428063"/>
          <action application="set" data="et_routing_host=10.0.0.12"/>
          <action application="set"
data="et_random_key=029645322327599477"/>
          <action application="set"
data="et_routed_number=555554351428063"/>
          <action application="set" data="et_routed_host=10.0.0.14"/>
          <action application="bridge" data="sofia/external/
555554351428063 at 10.0.0.14"/>
     </condition>
</extension>

If you want to offer other codecs you can add them to the codec-pref params.
They'll be listed in the preference order. FS will handle the transcoding
automatically.

You might also find it useful to read the Codec Negotiation page on the
Wiki, if you haven't already done so:
http://wiki.freeswitch.org/wiki/Codec_Negotiation

-Steve




On 14 June 2011 13:35, Gustavo Espeche <gustavo.espeche at easyipcall.com>wrote:

> Hi we are trying to do transcoding with freeswitch g711a-->g711u or
> g711-->speex but we can't do that FS do a trascoding, follow is a
> dialplan that we use with mod_curl
>
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> <section name="dialplan" description="Regex/XML Dialplan">
> <context name="public">
> <extension name="external">
> <condition field="destination_number" expression="^666654351428063">
> <action application="set" data="hangup_after_bridge=true"/>
> <action application="set" data="progress_timeout=20"/>
> <action application="set"
>
> data="export_vars=et_routing_number,et_routing_host,et_routed_number,et_routed_host,et_random_key"/>
> <action application="set" data="et_routing_number=666654351428063"/>
> <action application="set" data="et_routing_host=10.0.0.12"/>
> <action application="set" data="et_random_key=029645322327599477"/>
> <action application="set" data="proxy_media=false"/>
> <action application="set" data="bypass_media=true"/>
> <action application="set" data="absolute_codec_string=PCMU"/>
> <action application="export" data="nolocal:absolute_codec_string=PCMA"/>
> <action application="set" data="et_routed_number=555554351428063"/>
> <action application="set" data="et_routed_host=10.0.0.14"/>
> <action application="bridge"
> data="sofia/external/555554351428063 at 10.0.0.14"/>
> </condition>
> </extension>
> </context>
> </section>
> </document>
>
>  we relay can't understand what are we doing wrong.
> are we need configure someone special in FS? we are using
> sip_external_profile with port 5060.
>
>
> Regards.
>
> --
>
> Gustavo Espeche
> EasyIpCall S.R.L.
> www.easyipcall.com
> Bv Mitre 517 24? E
> Cordoba - Argentina
> Te: +54 - 351 - 4280633
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sid.kshatriya at gmail.com  Tue Jun 14 17:48:22 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Tue, 14 Jun 2011 19:18:22 +0530
Subject: [Freeswitch-users] Freeswitch transcoding
In-Reply-To: <BANLkTik1MXW6DcNYk3GbExR=w_6n623wTg@mail.gmail.com>
References: <1308054926.3665.6.camel@gustavo-laptop>
	<BANLkTik1MXW6DcNYk3GbExR=w_6n623wTg@mail.gmail.com>
Message-ID: <BANLkTimxK5ExV53zMJjogFEYvgNnSoODFw@mail.gmail.com>

Nice answer!

On Tue, Jun 14, 2011 at 6:44 PM, Steven Ayre <steveayre at gmail.com> wrote:

> At a guess, it's this line:
>
> <action application="set" data="bypass_media=true"/>
> Unless the media is going through FS it can't trancode. The SDP (codec
> list) will be forwarded from the aleg to the bleg so absolute_codec_string
> would be ignored.
>
> Also this has no effect, unless have proxy_media set to true on the sip
> profile. You don't need to set them to false, they'll already be false.
> You're correct to have this disabled though, as that would also prevent
> transcoding.
>
> <action application="set" data="proxy_media=false"/>
>
> This won't do as you think:
>
> <action application="set" data="absolute_codec_string=PCMU"/>
> It chooses the codec before you get to the dialplan. absolute_codec_string
> will only apply to the bleg. The incoming codec will either be picked from
> the inbound-codec-prefs param on the sip profile that receives the a-leg, or
> if you're using late-negotiation it'll pick that after the bridge so that
> it'll try to use the same codec as the bleg if possible and pick one of the
> other choices and transcode if that's not possible. absolute_codec_string
> overrides the outbound-codec-prefs sip profile param on the bleg.
>
> Try these on your sip profile:
>  <param name="inbound-codec-prefs" value="PCMU" />
>  <param name="outbound-codec-prefs" value="PCMA" />
>
> Then this will transcode:
>
> <extension name="external">
>      <condition field="destination_number" expression="^666654351428063">
>           <action application="set" data="hangup_after_bridge=true"/>
>           <action application="set" data="progress_timeout=20"/>
>           <action application="set"
> data="export_vars=et_routing_number,et_routing_host,et_routed_number,et_routed_host,et_random_key"/>
>           <action application="set"
> data="et_routing_number=666654351428063"/>
>           <action application="set" data="et_routing_host=10.0.0.12"/>
>           <action application="set"
> data="et_random_key=029645322327599477"/>
>           <action application="set"
> data="et_routed_number=555554351428063"/>
>           <action application="set" data="et_routed_host=10.0.0.14"/>
>           <action application="bridge" data="sofia/external/
> 555554351428063 at 10.0.0.14"/>
>      </condition>
> </extension>
>
> If you want to offer other codecs you can add them to the codec-pref
> params. They'll be listed in the preference order. FS will handle the
> transcoding automatically.
>
> You might also find it useful to read the Codec Negotiation page on the
> Wiki, if you haven't already done so:
> http://wiki.freeswitch.org/wiki/Codec_Negotiation
>
> -Steve
>
>
>
>
>
> On 14 June 2011 13:35, Gustavo Espeche <gustavo.espeche at easyipcall.com>wrote:
>
>> Hi we are trying to do transcoding with freeswitch g711a-->g711u or
>> g711-->speex but we can't do that FS do a trascoding, follow is a
>> dialplan that we use with mod_curl
>>
>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>> <document type="freeswitch/xml">
>> <section name="dialplan" description="Regex/XML Dialplan">
>> <context name="public">
>> <extension name="external">
>> <condition field="destination_number" expression="^666654351428063">
>> <action application="set" data="hangup_after_bridge=true"/>
>> <action application="set" data="progress_timeout=20"/>
>> <action application="set"
>>
>> data="export_vars=et_routing_number,et_routing_host,et_routed_number,et_routed_host,et_random_key"/>
>> <action application="set" data="et_routing_number=666654351428063"/>
>> <action application="set" data="et_routing_host=10.0.0.12"/>
>> <action application="set" data="et_random_key=029645322327599477"/>
>> <action application="set" data="proxy_media=false"/>
>> <action application="set" data="bypass_media=true"/>
>> <action application="set" data="absolute_codec_string=PCMU"/>
>> <action application="export" data="nolocal:absolute_codec_string=PCMA"/>
>> <action application="set" data="et_routed_number=555554351428063"/>
>> <action application="set" data="et_routed_host=10.0.0.14"/>
>> <action application="bridge"
>> data="sofia/external/555554351428063 at 10.0.0.14"/>
>> </condition>
>> </extension>
>> </context>
>> </section>
>> </document>
>>
>>  we relay can't understand what are we doing wrong.
>> are we need configure someone special in FS? we are using
>> sip_external_profile with port 5060.
>>
>>
>> Regards.
>>
>> --
>>
>> Gustavo Espeche
>> EasyIpCall S.R.L.
>> www.easyipcall.com
>> Bv Mitre 517 24? E
>> Cordoba - Argentina
>> Te: +54 - 351 - 4280633
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sidharth Kshatriya
www.sidk.info
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From ankitwalia4u at gmail.com  Tue Jun 14 18:06:01 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Tue, 14 Jun 2011 19:36:01 +0530
Subject: [Freeswitch-users] Adding a New User
In-Reply-To: <BANLkTinhiMjEDaM_okCP1f3H1VifvEmOfA@mail.gmail.com>
References: <BANLkTi=QJ7sAgP3dj6Zj-R7zQoLYZaoKdA@mail.gmail.com>
	<BANLkTinhiMjEDaM_okCP1f3H1VifvEmOfA@mail.gmail.com>
Message-ID: <BANLkTimX8tDdjbE0sJuKSOZ7yCn5tgh5sA@mail.gmail.com>

Thanks for responding Christian!!!

I am only having one network device.

Just to try my luck, I even tried setting the local_ip_v4 after your
suggestion. Still, the same issue :-(

Thanks
Ankit

On Tue, Jun 14, 2011 at 6:12 PM, Christian Benke <benkokakao at gmail.com>wrote:

> > freeswitch at ankit-PC> 2011-06-14 13:37:45.695430 [WARNING]
> sofia_reg.c:1872
> > Can't
> >  find user [1100 at 192.168.1.5]
> > You must define a domain called '192.168.1.5' in your directory and add a
> > user w
> > ith the id="1100" attribute
> > and you must configure your device to use the proper domain in it's
> > authenticati
> > on credentials.
> > 2011-06-14 13:37:45.703430 [WARNING] sofia_reg.c:1030 SIP auth failure
> > (REGISTER
> > ) on sofia profile 'internal' for [1100 at 192.168.1.5] from ip 192.168.1.5
> > I reloaded the XML also.
>
> Hi Ankit!
>
> Ignore the X-Lite message - this is a (minor) configuration error in
> FreeSWITCH.
>
> Is there more than one network-device configured on the machine?
>
> If yes, open /usr/local/freeswitch/conf/vars.xml and verify that you
> have the following settings somewhere:
>
> <!--IP Adress-->
> <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.1.5"/>
>
> <!-- Domain -->
> <!-- Sets the default domain. -->
> <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
> <X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/>
>
> Best regards
> Christian
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Life is like a rose its upto u feel it as its fragrance or thorns
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From benkokakao at gmail.com  Tue Jun 14 18:21:19 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Tue, 14 Jun 2011 16:21:19 +0200
Subject: [Freeswitch-users] Adding a New User
In-Reply-To: <BANLkTimX8tDdjbE0sJuKSOZ7yCn5tgh5sA@mail.gmail.com>
References: <BANLkTi=QJ7sAgP3dj6Zj-R7zQoLYZaoKdA@mail.gmail.com>
	<BANLkTinhiMjEDaM_okCP1f3H1VifvEmOfA@mail.gmail.com>
	<BANLkTimX8tDdjbE0sJuKSOZ7yCn5tgh5sA@mail.gmail.com>
Message-ID: <BANLkTin_L3-r0j4AvKH6=cU51ANm09aBsQ@mail.gmail.com>

On 14 June 2011 16:06, ankIT WALiA <ankitwalia4u at gmail.com> wrote:
> Thanks for responding Christian!!!
>
> I am only having one network device.
>
> Just to try my luck, I even tried setting the local_ip_v4 after your
> suggestion. Still, the same issue :-(

Add a cidr-parameter to the user-configuration in
conf/directory/default/ like that:

http://wiki.freeswitch.org/wiki/Acl#Users


Background:

conf/sip_profiles/internal.xml is the sip-profile authenticating the
directory-users, there you have a setting

<param name="apply-inbound-acl" value="domains"/>

The "domains"-acl is set in conf/autoload_configs/acl.conf.xml:

<list name="domains" default="deny">
      <node type="allow" domain="$${domain}"/>
</list>

Best regards,
Christian


From ankitwalia4u at gmail.com  Tue Jun 14 18:56:31 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Tue, 14 Jun 2011 20:26:31 +0530
Subject: [Freeswitch-users] Invalid Profile [external]
In-Reply-To: <BANLkTi=BxWtAbiBGnfAtu+x2U2qhiGZDbQ@mail.gmail.com>
References: <BANLkTinMcV_hY0yrg-FeD+4Aag+u+oB_=w@mail.gmail.com>
	<BANLkTimqAa=4+m6ni6Exac5xX9fU8eSiKA@mail.gmail.com>
	<BANLkTi=BxWtAbiBGnfAtu+x2U2qhiGZDbQ@mail.gmail.com>
Message-ID: <BANLkTinAjO4YOt9y5vMR1q0c0s244EBD8A@mail.gmail.com>

Christian,

I tried commenting out some code one after another, if there is any typo.
I got the error Invalid profile every time. I even tried removing iptel.xml
and add the content in external.xml, still issue persist.

Even for default configuration, this error seems to be coming.

Thanks
Ankit

On Tue, Jun 14, 2011 at 6:40 PM, Christian Benke <benkokakao at gmail.com>wrote:

> On 14 June 2011 14:51, Christian Benke <benkokakao at gmail.com> wrote:
> >> Now, after launching FS CLI, I run a "sofia profile external restart
> >> reloadxml" which is giving me message "Invalid Profile [external]"
> >
> > Please give me the full output of the command above, i believe it says
> > more than just "Invalid Profile".
>
> Oh! It's indeed just one line saying "2011-06-14 13:07:47.335914
> [WARNING] sofia.c:4044 No Such Profile 'external'"
>
> You probably have a typo in your sip-profile-configuration, in case
> you didn't change it intentionally.
>
> The default configuration for the external sip-profile is
> /usr/local/freeswitch/conf/sip_profiles/external.xml(Modify the path
> according to your Win7-installation).
>
> Please verify you have a section in your configuration that says:
>
> <profile name="external">
>  <gateways>
>    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
>  </gateways>
>
> ...lots of other stuff in default-config...
>
> </profile>
>
>
> Then you add the iptel-configuration you've posted above in
> /usr/local/freeswitch/conf/sip_profiles/external/iptel.xml(If this
> didn't happen already)
>
> Best regards,
> Christian
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Life is like a rose its upto u feel it as its fragrance or thorns
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From henk at oegema.com  Tue Jun 14 14:50:31 2011
From: henk at oegema.com (Henk Oegema)
Date: Tue, 14 Jun 2011 12:50:31 +0200
Subject: [Freeswitch-users] One way audio with Localphone .com
Message-ID: <1308048631.2133.76.camel@DELL>

I'm in the process of changing from Asterisk to Freeswitch (Version
1.0.head (git-0675b59 2011-06-06 21-28-14 -0500)).
I have a problem with one way audio, when using Localphone.com with FS.
(Audio is OK both way with localphone.com in Asterisk). (Audio is also
OK bothway in FS with powervoip.com and jumblo.com)

Problem:
mobile -> Localphone.com -> FS   audio is OK
FS -> Localphone.com -> mobile   audio NOK

My setup is a little bit specific :  internet -> internetmodem ->
(ip:62.195.45.192) router1 -> switch ->(ip:192.1.68.1.144) router2 ->
(ip:10.0.0.101) FS server
Explanation of above line;
router1 has external-ip:62.195.45.192
router1 gives router2  external-ip:192.1.68.1.144
router2 gives FS server internal-ip:10.0.0.101

When I connect to localphone.com with Asterisk, I see:
Device name: Asterisk PBX 1.6.2.7
Status: Online
Received IP address: 62.195.45.192
Contact IP address: 62.195.45.192  <----------!!

When I connect to localphone.com with FS, I see:
Device name: FreeSWITCH-mod_sofia/1.0.head-git-0675b59.....
Received IP address: 62.195.45.192
Contact IP address: 192.168.1.144  <-----------??

It looks to me that the one way audio has to do with the Contact IP
address:192.168.1.144 (???)
Is it possible to change a setting in FS so that Localphone.com will get
Contact IP address: 62.195.45.192 ? (may be this solves the problem)

(I don't think (?) it has to do with port-forwarding, since all audio is
ok with Asterisk)

Rgds.










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From ankitwalia4u at gmail.com  Tue Jun 14 19:05:11 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Tue, 14 Jun 2011 20:35:11 +0530
Subject: [Freeswitch-users] Invalid Profile [external]
In-Reply-To: <BANLkTinAjO4YOt9y5vMR1q0c0s244EBD8A@mail.gmail.com>
References: <BANLkTinMcV_hY0yrg-FeD+4Aag+u+oB_=w@mail.gmail.com>
	<BANLkTimqAa=4+m6ni6Exac5xX9fU8eSiKA@mail.gmail.com>
	<BANLkTi=BxWtAbiBGnfAtu+x2U2qhiGZDbQ@mail.gmail.com>
	<BANLkTinAjO4YOt9y5vMR1q0c0s244EBD8A@mail.gmail.com>
Message-ID: <BANLkTin+kELEx-ZCr8+Scq4RRu6e6RKofg@mail.gmail.com>

Where I can modify the path for the default configuration for the external
sip-profile is
/usr/local/freeswitch/conf/
sip_profiles/external.xml(Modify the path
according to your Win7-installation).

I have seen, everywhere there is a relative path.


On Tue, Jun 14, 2011 at 8:26 PM, ankIT WALiA <ankitwalia4u at gmail.com> wrote:

> Christian,
>
> I tried commenting out some code one after another, if there is any typo.
> I got the error Invalid profile every time. I even tried removing iptel.xml
> and add the content in external.xml, still issue persist.
>
> Even for default configuration, this error seems to be coming.
>
> Thanks
> Ankit
>
>
> On Tue, Jun 14, 2011 at 6:40 PM, Christian Benke <benkokakao at gmail.com>wrote:
>
>> On 14 June 2011 14:51, Christian Benke <benkokakao at gmail.com> wrote:
>> >> Now, after launching FS CLI, I run a "sofia profile external restart
>> >> reloadxml" which is giving me message "Invalid Profile [external]"
>> >
>> > Please give me the full output of the command above, i believe it says
>> > more than just "Invalid Profile".
>>
>> Oh! It's indeed just one line saying "2011-06-14 13:07:47.335914
>> [WARNING] sofia.c:4044 No Such Profile 'external'"
>>
>> You probably have a typo in your sip-profile-configuration, in case
>> you didn't change it intentionally.
>>
>> The default configuration for the external sip-profile is
>> /usr/local/freeswitch/conf/sip_profiles/external.xml(Modify the path
>> according to your Win7-installation).
>>
>> Please verify you have a section in your configuration that says:
>>
>> <profile name="external">
>>  <gateways>
>>    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
>>  </gateways>
>>
>> ...lots of other stuff in default-config...
>>
>> </profile>
>>
>>
>> Then you add the iptel-configuration you've posted above in
>> /usr/local/freeswitch/conf/sip_profiles/external/iptel.xml(If this
>> didn't happen already)
>>
>> Best regards,
>> Christian
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Life is like a rose its upto u feel it as its fragrance or thorns
>



-- 
Life is like a rose its upto u feel it as its fragrance or thorns
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From wes-fs at 499x.com  Tue Jun 14 19:06:05 2011
From: wes-fs at 499x.com (wes-fs at 499x.com)
Date: Tue, 14 Jun 2011 10:06:05 -0500
Subject: [Freeswitch-users] using fs api to originate a call and record
 it
In-Reply-To: <BANLkTik6Wh7vccJrCx7rUoKSJyYcAzzr3g@mail.gmail.com>
References: <4DF278F9.6000201@499x.com>
	<BANLkTik6Wh7vccJrCx7rUoKSJyYcAzzr3g@mail.gmail.com>
Message-ID: <4DF778DD.7060805@499x.com>

This did not work, I still heard the ringing.

On 6/10/2011 5:30 PM, Michael Collins wrote:
> Try this:
>
> originate 
> {media_bug_answer_req=true}sofia/sipinterface_1/912625551212 at 141.xxx.xxx.xxx 
> &record(/tmp/myrecording.wav)
>
> Let us know if it works.
> -MC
>
> On Fri, Jun 10, 2011 at 1:05 PM, <wes-fs at 499x.com 
> <mailto:wes-fs at 499x.com>> wrote:
>
>     I'm extremely new to freeswitch, and anything like it, so please
>     forgive me!
>
>     I'd like to be able to use the API to originate a call and record it.
>     So far, I have the following figured out:
>
>     originate sofia/sipinterface_1/912625551212 at 141.xxx.xxx.xxx
>     &record(/tmp/myrecording.wav)
>     And it dials out to an external number and successfully records the
>     conversation.
>
>     But, it also records the ringing.  When I was trying this through a
>     dialplan, I found an option:
>     <action application="set" data="media_bug_answer_req=true"/>
>     which delayed the recording until the call was answered.
>
>     Is there a way to do the same thing through the API?
>
>     Thanks!
>     Wes
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From msc at freeswitch.org  Tue Jun 14 19:18:53 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 14 Jun 2011 08:18:53 -0700
Subject: [Freeswitch-users] Invalid Profile [external]
In-Reply-To: <BANLkTin+kELEx-ZCr8+Scq4RRu6e6RKofg@mail.gmail.com>
References: <BANLkTinMcV_hY0yrg-FeD+4Aag+u+oB_=w@mail.gmail.com>
	<BANLkTimqAa=4+m6ni6Exac5xX9fU8eSiKA@mail.gmail.com>
	<BANLkTi=BxWtAbiBGnfAtu+x2U2qhiGZDbQ@mail.gmail.com>
	<BANLkTinAjO4YOt9y5vMR1q0c0s244EBD8A@mail.gmail.com>
	<BANLkTin+kELEx-ZCr8+Scq4RRu6e6RKofg@mail.gmail.com>
Message-ID: <BANLkTinWjFsoFi-o526EY6G=fDLgobdOXw@mail.gmail.com>

Be sure that the external profile is actually coming up. Watch the log when
you start FS - if the external profile's port is already in use then the
profile will not start.
-MC

On Tue, Jun 14, 2011 at 8:05 AM, ankIT WALiA <ankitwalia4u at gmail.com> wrote:

> Where I can modify the path for the default configuration for the external
> sip-profile is
> /usr/local/freeswitch/conf/
> sip_profiles/external.xml(Modify the path
> according to your Win7-installation).
>
> I have seen, everywhere there is a relative path.
>
>
> On Tue, Jun 14, 2011 at 8:26 PM, ankIT WALiA <ankitwalia4u at gmail.com>wrote:
>
>> Christian,
>>
>> I tried commenting out some code one after another, if there is any typo.
>> I got the error Invalid profile every time. I even tried removing
>> iptel.xml and add the content in external.xml, still issue persist.
>>
>> Even for default configuration, this error seems to be coming.
>>
>> Thanks
>> Ankit
>>
>>
>> On Tue, Jun 14, 2011 at 6:40 PM, Christian Benke <benkokakao at gmail.com>wrote:
>>
>>> On 14 June 2011 14:51, Christian Benke <benkokakao at gmail.com> wrote:
>>> >> Now, after launching FS CLI, I run a "sofia profile external restart
>>> >> reloadxml" which is giving me message "Invalid Profile [external]"
>>> >
>>> > Please give me the full output of the command above, i believe it says
>>> > more than just "Invalid Profile".
>>>
>>> Oh! It's indeed just one line saying "2011-06-14 13:07:47.335914
>>> [WARNING] sofia.c:4044 No Such Profile 'external'"
>>>
>>> You probably have a typo in your sip-profile-configuration, in case
>>> you didn't change it intentionally.
>>>
>>> The default configuration for the external sip-profile is
>>> /usr/local/freeswitch/conf/sip_profiles/external.xml(Modify the path
>>> according to your Win7-installation).
>>>
>>> Please verify you have a section in your configuration that says:
>>>
>>> <profile name="external">
>>>  <gateways>
>>>    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
>>>  </gateways>
>>>
>>> ...lots of other stuff in default-config...
>>>
>>> </profile>
>>>
>>>
>>> Then you add the iptel-configuration you've posted above in
>>> /usr/local/freeswitch/conf/sip_profiles/external/iptel.xml(If this
>>> didn't happen already)
>>>
>>> Best regards,
>>> Christian
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Life is like a rose its upto u feel it as its fragrance or thorns
>>
>
>
>
> --
> Life is like a rose its upto u feel it as its fragrance or thorns
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sid.kshatriya at gmail.com  Tue Jun 14 19:21:53 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Tue, 14 Jun 2011 20:51:53 +0530
Subject: [Freeswitch-users] Want to play music on hold while executing a
 time consuming operation in Lua
Message-ID: <BANLkTinXFZPeprzatd45N42ZuP2obvU5JQ@mail.gmail.com>

I am implementing an IVR using Lua in Freeswitch. In my Lua script I use
curl to a web service. Sometimes the response takes a long time to come
back. During that time period I would like to play music on hold.

I have searched the freeswitch discussion archives a lot. There seem to be
many suggested ways to implement music on hold from a Lua script but the
answers are not very clear / not really applicable to my use case. I don't
know what method I should use and which one is recommended.

*Method 1:* Transfer the call to 9664 (music on hold extension). However the
implementation for this solution for this does not seem to be available in
Lua. For example: How would I transfer the call back?
*Method 2:* Using bgapi uuid_park park the call and using uuid_broadcast
play an audio file. Again what do I do to unpark the call..?

Thanks,

Sidharth
-- 
Sidharth Kshatriya
www.sidk.info
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From wes-fs at 499x.com  Tue Jun 14 19:30:34 2011
From: wes-fs at 499x.com (wes-fs at 499x.com)
Date: Tue, 14 Jun 2011 10:30:34 -0500
Subject: [Freeswitch-users] using fs api to originate a call and record
 it
In-Reply-To: <4DF778DD.7060805@499x.com>
References: <4DF278F9.6000201@499x.com>	<BANLkTik6Wh7vccJrCx7rUoKSJyYcAzzr3g@mail.gmail.com>
	<4DF778DD.7060805@499x.com>
Message-ID: <4DF77E9A.6010701@499x.com>

based on the documentation here, it looks like I might be able to 
execute a script, instead of directly doing the record command.  Then 
the script could maybe set the parameters I need?

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



On 6/14/2011 10:06 AM, wes-fs at 499x.com wrote:
> This did not work, I still heard the ringing.
>
> On 6/10/2011 5:30 PM, Michael Collins wrote:
>> Try this:
>>
>> originate 
>> {media_bug_answer_req=true}sofia/sipinterface_1/912625551212 at 141.xxx.xxx.xxx 
>> &record(/tmp/myrecording.wav)
>>
>> Let us know if it works.
>> -MC
>>
>> On Fri, Jun 10, 2011 at 1:05 PM, <wes-fs at 499x.com 
>> <mailto:wes-fs at 499x.com>> wrote:
>>
>>     I'm extremely new to freeswitch, and anything like it, so please
>>     forgive me!
>>
>>     I'd like to be able to use the API to originate a call and record it.
>>     So far, I have the following figured out:
>>
>>     originate sofia/sipinterface_1/912625551212 at 141.xxx.xxx.xxx
>>     &record(/tmp/myrecording.wav)
>>     And it dials out to an external number and successfully records the
>>     conversation.
>>
>>     But, it also records the ringing.  When I was trying this through a
>>     dialplan, I found an option:
>>     <action application="set" data="media_bug_answer_req=true"/>
>>     which delayed the recording until the call was answered.
>>
>>     Is there a way to do the same thing through the API?
>>
>>     Thanks!
>>     Wes
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brad at tech21.com  Tue Jun 14 21:14:32 2011
From: brad at tech21.com (Brad Mina)
Date: Tue, 14 Jun 2011 10:14:32 -0700
Subject: [Freeswitch-users] One way audio with Localphone .com
In-Reply-To: <1308048631.2133.76.camel@DELL>
References: <1308048631.2133.76.camel@DELL>
Message-ID: <BANLkTinYLVq0xbmd4uwFxvdWqrJpE8OceQ@mail.gmail.com>

A double NAT has always been very tricky for me, I ended up merging my
doubled subnet into the primary.

As for your contact address, try using STUN to check your external IP in
your vars.xml and external.xml

http://wiki.freeswitch.org/wiki/NAT
<http://wiki.freeswitch.org/wiki/NAT>

On Tue, Jun 14, 2011 at 3:50 AM, Henk Oegema <henk at oegema.com> wrote:

>  I'm in the process of changing from Asterisk to Freeswitch (Version
> 1.0.head (git-0675b59 2011-06-06 21-28-14 -0500)).
> I have a problem with one way audio, when using Localphone.com with FS.
> (Audio is OK both way with localphone.com in Asterisk). (Audio is also OK
> bothway in FS with powervoip.com and jumblo.com)
>
> Problem:
> mobile -> Localphone.com -> FS   audio is OK
> FS -> Localphone.com -> mobile   audio NOK
>
> My setup is a little bit specific :  internet -> internetmodem ->
> (ip:62.195.45.192) router1 -> switch ->(ip:192.1.68.1.144) router2 ->
> (ip:10.0.0.101) FS server
> Explanation of above line;
> router1 has external-ip:62.195.45.192
> router1 gives router2  external-ip:192.1.68.1.144
> router2 gives FS server internal-ip:10.0.0.101
>
> When I connect to localphone.com with Asterisk, I see:
> Device name: Asterisk PBX 1.6.2.7
> Status: Online
> Received IP address: 62.195.45.192
> Contact IP address: 62.195.45.192  <----------!!
>
> When I connect to localphone.com with FS, I see:
> Device name: FreeSWITCH-mod_sofia/1.0.head-git-0675b59.....
> Received IP address: 62.195.45.192
> Contact IP address: 192.168.1.144  <-----------??
>
> It looks to me that the one way audio has to do with the Contact IP
> address:192.168.1.144 (???)
> Is it possible to change a setting in FS so that Localphone.com will get
> Contact IP address: 62.195.45.192 ? (may be this solves the problem)
>
> (I don't think (?) it has to do with port-forwarding, since all audio is ok
> with Asterisk)
>
> Rgds.
>
>
>
>
>
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From frankie.k.yiu at gmail.com  Tue Jun 14 21:54:32 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Tue, 14 Jun 2011 10:54:32 -0700
Subject: [Freeswitch-users] How to record incoming audio only (not both
	direction)?
Message-ID: <BANLkTiku+_e9aFgy8zfm7-1=7KcgpU4SQA@mail.gmail.com>

Hi,

I would like to record the incoming audio during a call when making a call
to a callee, so that we can optimize the voice mail detection algorithm when
debugging.

I know there is a function: "switch_ivr_record_session" to record the whole
session in both direction (incoming & outgoing) but I can not find a
function to record only the incoming audio.  Could someone please help me on
how I can record only the incoming direction audio?

Thanks,
Frankie
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From avi at avimarcus.net  Tue Jun 14 22:00:44 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 14 Jun 2011 21:00:44 +0300
Subject: [Freeswitch-users] How to record incoming audio only (not both
	direction)?
In-Reply-To: <BANLkTiku+_e9aFgy8zfm7-1=7KcgpU4SQA@mail.gmail.com>
References: <BANLkTiku+_e9aFgy8zfm7-1=7KcgpU4SQA@mail.gmail.com>
Message-ID: <BANLkTinP0kD=EJCYxNy8KR7JctkTGQS-jQ@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record
<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record>Check the
channel variable that you can set.. e.g.
RECORD_WRITE_ONLY<http://wiki.freeswitch.org/wiki/Variable_RECORD_WRITE_ONLY>

-Avi Marcus


On Tue, Jun 14, 2011 at 8:54 PM, Frankie Yiu <frankie.k.yiu at gmail.com>wrote:

> Hi,
>
> I would like to record the incoming audio during a call when making a call
> to a callee, so that we can optimize the voice mail detection algorithm when
> debugging.
>
> I know there is a function: "switch_ivr_record_session" to record the whole
> session in both direction (incoming & outgoing) but I can not find a
> function to record only the incoming audio.  Could someone please help me on
> how I can record only the incoming direction audio?
>
> Thanks,
> Frankie
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From justlikeef at gmail.com  Tue Jun 14 22:01:20 2011
From: justlikeef at gmail.com (Rob Hutton)
Date: Tue, 14 Jun 2011 14:01:20 -0400
Subject: [Freeswitch-users] Trying to get TLS/SRTP working
Message-ID: <BANLkTimLoVv7sRw-OOY-th0fC94uj-oYzw@mail.gmail.com>

Trying to get TLS working.  When sofia loads, I see that it seems to accept
the parameters:

http://pastebin.freeswitch.org/16487

But sofia status shows nothing running on the TLS ports:
                     Name
Type                                       Data      State
=================================================================================================
           sipinterface_3       profile
sip:mod_sofia at 192.168.2.25:5080      RUNNING (0)
           sipinterface_2       profile
sip:mod_sofia at 192.168.2.25:5070      RUNNING (0)
             192.168.2.25         alias
sipinterface_1      ALIASED
           sipinterface_1       profile
sip:mod_sofia at 192.168.2.25:5060      RUNNING (0)
              voicemail_1         alias
sipinterface_1      ALIASED
=================================================================================================
3 profiles 2 aliases


/usr/local/freeswitch/conf/ssl looks like:
-rw-r--r-- 1 root root 3627 Jun 14 12:55 agent.pem
-rw-r--r-- 1 root root 1996 Jun 14 13:04 cafile.pem


Freeswitch is compiled with SSL support:

ldd freeswitch
        linux-vdso.so.1 =>  (0x00007fff415ff000)
        libm.so.6 => /lib64/libm.so.6 (0x00007fab636b8000)
        libfreeswitch.so.1 => /usr/local/freeswitch/lib/libfreeswitch.so.1
(0x00007fab632c1000)
        libuuid.so.1 => /lib64/libuuid.so.1 (0x00007fab630bc000)
        librt.so.1 => /lib64/librt.so.1 (0x00007fab62eb3000)
        libcrypt.so.1 => /lib64/libcrypt.so.1 (0x00007fab62c78000)
        libpthread.so.0 => /lib64/libpthread.so.0 (0x00007fab62a5b000)
        libssl.so.1.0.0 => /lib64/libssl.so.1.0.0 (0x00007fab627ff000)
        libcrypto.so.1.0.0 => /lib64/libcrypto.so.1.0.0 (0x00007fab6244e000)
        libdl.so.2 => /lib64/libdl.so.2 (0x00007fab6224a000)
        libz.so.1 => /lib64/libz.so.1 (0x00007fab62032000)
        libncurses.so.5 => /lib64/libncurses.so.5 (0x00007fab61ddd000)
        libc.so.6 => /lib64/libc.so.6 (0x00007fab61a70000)
        libstdc++.so.6 => /usr/lib64/libstdc++.so.6 (0x00007fab61767000)
        libgcc_s.so.1 => /lib64/libgcc_s.so.1 (0x00007fab61551000)
        libodbc.so.1 => /usr/lib64/libodbc.so.1 (0x00007fab612e5000)
        /lib64/ld-linux-x86-64.so.2 (0x00007fab6390f000)
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From gerrit308 at gmail.com  Wed Jun 15 00:05:24 2011
From: gerrit308 at gmail.com (humbr)
Date: Tue, 14 Jun 2011 13:05:24 -0700 (PDT)
Subject: [Freeswitch-users] Freeswitch.dbh and affected_rows() question
Message-ID: <1308081924858-6475723.post@n2.nabble.com>

Basically, does this function do anything if I am using core:db? In my test
code it consistently returns 0 even though the just completed query did
return 1 or more rows.

I couldn't find this function in switch_core_db.c

Gerrit

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-dbh-and-affected-rows-question-tp6475723p6475723.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From wes-fs at 499x.com  Wed Jun 15 01:48:11 2011
From: wes-fs at 499x.com (Wes)
Date: Tue, 14 Jun 2011 16:48:11 -0500
Subject: [Freeswitch-users] invalid application lua
Message-ID: <4DF7D71B.3020103@499x.com>

I'm getting "invalid application lua" however, in modules.conf, the line 
for <load module="mod_lua"/> is active and not commented out.

Any help is appreciated. Thanks.


From sos at sokhapkin.dyndns.org  Wed Jun 15 01:55:49 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 14 Jun 2011 17:55:49 -0400
Subject: [Freeswitch-users] invalid application lua
In-Reply-To: <4DF7D71B.3020103@499x.com>
References: <4DF7D71B.3020103@499x.com>
Message-ID: <201106141755.49546.sos@sokhapkin.dyndns.org>

Is mod_lua compiled and installed?

On Tuesday 14 June 2011, Wes wrote:
> I'm getting "invalid application lua" however, in modules.conf, the line
> for <load module="mod_lua"/> is active and not commented out.
> 
> Any help is appreciated. Thanks.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From avi at avimarcus.net  Wed Jun 15 01:55:51 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 15 Jun 2011 00:55:51 +0300
Subject: [Freeswitch-users] invalid application lua
In-Reply-To: <4DF7D71B.3020103@499x.com>
References: <4DF7D71B.3020103@499x.com>
Message-ID: <BANLkTim01AOjuyx8jgCuaF4ki_khnr0UUg@mail.gmail.com>

What happens when you run "reload mod_lua" from the CLI?

 -Avi Marcus


On Wed, Jun 15, 2011 at 12:48 AM, Wes <wes-fs at 499x.com> wrote:

> I'm getting "invalid application lua" however, in modules.conf, the line
> for <load module="mod_lua"/> is active and not commented out.
>
> Any help is appreciated. Thanks.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From henk at oegema.com  Wed Jun 15 02:00:11 2011
From: henk at oegema.com (Henk Oegema)
Date: Wed, 15 Jun 2011 00:00:11 +0200
Subject: [Freeswitch-users] One way audio with Localphone .com
In-Reply-To: <1308048631.2133.76.camel@DELL>
References: <1308048631.2133.76.camel@DELL>
Message-ID: <1308088811.2247.8.camel@DELL>

Surprisingly it suddenly started to work.
I'm sorry that I can't explain why! 
I tried so many things, that I lost control of what I did.

The logon settings are still the same:
Device name: FreeSWITCH-mod_sofia/1.0.head-git-0675b59.....
Received IP address: 62.195.45.192
Contact IP address: 192.168.1.144  <-----------Now it works !! 



On Tue, 2011-06-14 at 12:50 +0200, Henk Oegema wrote:

> I'm in the process of changing from Asterisk to Freeswitch (Version
> 1.0.head (git-0675b59 2011-06-06 21-28-14 -0500)).
> I have a problem with one way audio, when using Localphone.com with
> FS. (Audio is OK both way with localphone.com in Asterisk). (Audio is
> also OK bothway in FS with powervoip.com and jumblo.com)
> 
> Problem:
> mobile -> Localphone.com -> FS   audio is OK
> FS -> Localphone.com -> mobile   audio NOK
> 
> My setup is a little bit specific :  internet -> internetmodem ->
> (ip:62.195.45.192) router1 -> switch ->(ip:192.1.68.1.144) router2 ->
> (ip:10.0.0.101) FS server
> Explanation of above line;
> router1 has external-ip:62.195.45.192
> router1 gives router2  external-ip:192.1.68.1.144
> router2 gives FS server internal-ip:10.0.0.101
> 
> When I connect to localphone.com with Asterisk, I see:
> Device name: Asterisk PBX 1.6.2.7
> Status: Online
> Received IP address: 62.195.45.192
> Contact IP address: 62.195.45.192  <----------!!
> 
> When I connect to localphone.com with FS, I see:
> Device name: FreeSWITCH-mod_sofia/1.0.head-git-0675b59.....
> Received IP address: 62.195.45.192
> Contact IP address: 192.168.1.144  <-----------??
> 
> It looks to me that the one way audio has to do with the Contact IP
> address:192.168.1.144 (???)
> Is it possible to change a setting in FS so that Localphone.com will
> get Contact IP address: 62.195.45.192 ? (may be this solves the
> problem)
> 
> (I don't think (?) it has to do with port-forwarding, since all audio
> is ok with Asterisk)
> 
> Rgds.
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
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From paul at iamfine.com  Wed Jun 15 00:41:35 2011
From: paul at iamfine.com (paul at iamfine.com)
Date: Tue, 14 Jun 2011 21:41:35 +0100
Subject: [Freeswitch-users] failure to start a session from within lua for
	outbound call
Message-ID: <DE04BDDC-761E-484C-938A-A7FE598F3988@iamfine.com>

Having problems getting a session created from a lua script - its the latest build on a centos system

this is the lua code 

-----------------------
freeswitch.console_log("info", "Lua in da house!!!\n");

local session = freeswitch.Session("sofia/00.00.127.61/1000");
session:execute("playback", "/sr8k.wav");
session:hangup();
-----------------------

***** please note i changed the ip address to protect its identity *****
**********




please help - i am tearing my hair out on this one

Paul


+OK

freeswitch at internal> 2011-06-14 20:31:01.476406 [INFO] switch_cpp.cpp:1197 Lua in da house!!!
2011-06-14 20:31:01.476406 [DEBUG] switch_ivr_originate.c:1869 Parsing global variables
2011-06-14 20:31:01.476406 [NOTICE] switch_channel.c:833 New Channel sofia/internal/1000 [c9c5d8ec-af7f-4153-a844-5000983170d5]
2011-06-14 20:31:01.476406 [DEBUG] mod_sofia.c:4392 (sofia/internal/1000) State Change CS_NEW -> CS_INIT
2011-06-14 20:31:01.476406 [DEBUG] switch_core_session.c:1118 Send signal sofia/internal/1000 [BREAK]
2011-06-14 20:31:01.476406 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1000) Running State Change CS_INIT
2011-06-14 20:31:01.476406 [DEBUG] switch_core_state_machine.c:361 (sofia/internal/1000) State INIT
2011-06-14 20:31:01.476406 [DEBUG] mod_sofia.c:85 sofia/internal/1000 SOFIA INIT
2011-06-14 20:31:01.476406 [DEBUG] mod_sofia.c:125 (sofia/internal/1000) State Change CS_INIT -> CS_ROUTING
2011-06-14 20:31:01.476406 [DEBUG] switch_core_session.c:1118 Send signal sofia/internal/1000 [BREAK]
2011-06-14 20:31:01.476406 [DEBUG] switch_core_state_machine.c:361 (sofia/internal/1000) State INIT going to sleep
2011-06-14 20:31:01.476406 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1000) Running State Change CS_ROUTING
2011-06-14 20:31:01.476406 [DEBUG] switch_channel.c:1736 (sofia/internal/1000) Callstate Change DOWN -> RINGING
2011-06-14 20:31:01.476406 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/1000) State ROUTING
2011-06-14 20:31:01.476406 [DEBUG] mod_sofia.c:148 sofia/internal/1000 SOFIA ROUTING
2011-06-14 20:31:01.476406 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/1000) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2011-06-14 20:31:01.476406 [DEBUG] switch_core_session.c:1118 Send signal sofia/internal/1000 [BREAK]
2011-06-14 20:31:01.476406 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/1000) State ROUTING going to sleep
2011-06-14 20:31:01.476406 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1000) Running State Change CS_CONSUME_MEDIA
2011-06-14 20:31:01.476406 [DEBUG] switch_core_state_machine.c:383 (sofia/internal/1000) State CONSUME_MEDIA
2011-06-14 20:31:01.476406 [DEBUG] switch_core_state_machine.c:383 (sofia/internal/1000) State CONSUME_MEDIA going to sleep
2011-06-14 20:31:01.476406 [DEBUG] sofia.c:4800 Channel sofia/internal/1000 entering state [calling][0]
2011-06-14 20:31:01.638509 [DEBUG] sofia.c:4800 Channel sofia/internal/1000 entering state [terminated][503]
2011-06-14 20:31:01.638509 [DEBUG] switch_channel.c:2641 (sofia/internal/1000) Callstate Change RINGING -> HANGUP
2011-06-14 20:31:01.638509 [NOTICE] sofia.c:5522 Hangup sofia/internal/1000 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
2011-06-14 20:31:01.638509 [DEBUG] switch_ivr_originate.c:3308 Originate Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE]
2011-06-14 20:31:01.638509 [ERR] switch_cpp.cpp:655 session is not initalized
2011-06-14 20:31:01.638509 [ERR] switch_cpp.cpp:617 session is not initalized
2011-06-14 20:31:01.638509 [DEBUG] switch_channel.c:2657 Send signal sofia/internal/1000 [KILL]
2011-06-14 20:31:01.638509 [DEBUG] switch_core_session.c:1118 Send signal sofia/internal/1000 [BREAK]
2011-06-14 20:31:01.638509 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1000) Running State Change CS_HANGUP
2011-06-14 20:31:01.638509 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/1000) State HANGUP
2011-06-14 20:31:01.638509 [DEBUG] mod_sofia.c:452 sofia/internal/1000 Overriding SIP cause 503 with 503 from the other leg
2011-06-14 20:31:01.638509 [DEBUG] mod_sofia.c:458 Channel sofia/internal/1000 hanging up, cause: NORMAL_TEMPORARY_FAILURE
2011-06-14 20:31:01.638509 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1000 Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE
2011-06-14 20:31:01.638509 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/1000) State HANGUP going to sleep
2011-06-14 20:31:01.638509 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/1000) State Change CS_HANGUP -> CS_REPORTING
2011-06-14 20:31:01.638509 [DEBUG] switch_core_session.c:1118 Send signal sofia/internal/1000 [BREAK]
2011-06-14 20:31:01.638509 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1000) Running State Change CS_REPORTING
2011-06-14 20:31:01.638509 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/1000) State REPORTING
2011-06-14 20:31:02.631557 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1000 Standard REPORTING, cause: NORMAL_TEMPORARY_FAILURE
2011-06-14 20:31:02.631557 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/1000) State REPORTING going to sleep
2011-06-14 20:31:02.631557 [DEBUG] switch_core_state_machine.c:350 (sofia/internal/1000) State Change CS_REPORTING -> CS_DESTROY
2011-06-14 20:31:02.631557 [DEBUG] switch_core_session.c:1118 Send signal sofia/internal/1000 [BREAK]
2011-06-14 20:31:02.631557 [DEBUG] switch_core_session.c:1290 Session 8 (sofia/internal/1000) Locked, Waiting on external entities
2011-06-14 20:31:02.631557 [NOTICE] switch_core_session.c:1308 Session 8 (sofia/internal/1000) Ended
2011-06-14 20:31:02.631557 [NOTICE] switch_core_session.c:1310 Close Channel sofia/internal/1000 [CS_DESTROY]
2011-06-14 20:31:02.631557 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/1000) Callstate Change HANGUP -> DOWN
2011-06-14 20:31:02.631557 [DEBUG] switch_core_state_machine.c:457 (sofia/internal/1000) Running State Change CS_DESTROY
2011-06-14 20:31:02.631557 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1000) State DESTROY
2011-06-14 20:31:02.631557 [DEBUG] mod_sofia.c:363 sofia/internal/1000 SOFIA DESTROY
2011-06-14 20:31:02.631557 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1000 Standard DESTROY
2011-06-14 20:31:02.631557 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1000) State DESTROY going to sleep

From marketing at cluecon.com  Wed Jun 15 02:21:46 2011
From: marketing at cluecon.com (marketing at cluecon.com)
Date: Tue, 14 Jun 2011 22:21:46 +0000
Subject: [Freeswitch-users] ClueCon 2011 is Almost Here!
Message-ID: <00000130903f02de-584b4a90-af91-4575-a025-cd0787a6c37a-000000@email.amazonses.com>

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From henk at oegema.com  Wed Jun 15 02:33:33 2011
From: henk at oegema.com (Henk Oegema)
Date: Wed, 15 Jun 2011 00:33:33 +0200
Subject: [Freeswitch-users] Warning - SIP auth challenge
Message-ID: <1308090813.2247.14.camel@DELL>

What is the meaning of getting (regular) this message:
.....[WARNING] sofia_reg.c:1241 SIP auth challenge (REGISTER) on sofia
profile 'internal' for [1001 at 10.0.0.101] from ip 10.0.0.12 ??

Extension 1001 is a Portech GSM gateway MV370. 
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From avi at avimarcus.net  Wed Jun 15 02:38:29 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 15 Jun 2011 01:38:29 +0300
Subject: [Freeswitch-users] Warning - SIP auth challenge
In-Reply-To: <1308090813.2247.14.camel@DELL>
References: <1308090813.2247.14.camel@DELL>
Message-ID: <BANLkTi=xGm2UwRDwJvZoKu_1ssUW82T3=Q@mail.gmail.com>

It's just a notice that there's a register attempt. If it's every 30 or 60
minutes from a known extension, it's fine.

It's there for fail2ban to notice excessive register attempts so it can
block abuse from clients that try to register but don't submit a password to
avoid setting off the warning bells.

-Avi


On Wed, Jun 15, 2011 at 1:33 AM, Henk Oegema <henk at oegema.com> wrote:

>  What is the meaning of getting (regular) this message:
> .....[WARNING] sofia_reg.c:1241 SIP auth challenge (REGISTER) on sofia
> profile 'internal' for [1001 at 10.0.0.101] from ip 10.0.0.12 ??
>
> Extension 1001 is a Portech GSM gateway MV370.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From henk at oegema.com  Wed Jun 15 02:48:06 2011
From: henk at oegema.com (Henk Oegema)
Date: Wed, 15 Jun 2011 00:48:06 +0200
Subject: [Freeswitch-users] [Fwd: Re:  Warning - SIP auth challenge]
Message-ID: <1308091686.2247.16.camel@DELL>

But the message is every 4 minutes.

-------- Forwarded Message --------
From: Avi Marcus <avi at avimarcus.net>
Reply-to: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Warning - SIP auth challenge
Date: Wed, 15 Jun 2011 01:38:29 +0300

It's just a notice that there's a register attempt. If it's every 30 or
60 minutes from a known extension, it's fine.


It's there for fail2ban to notice excessive register attempts so it can
block abuse from clients that try to register but don't submit a
password to avoid setting off the warning bells.

-Avi




On Wed, Jun 15, 2011 at 1:33 AM, Henk Oegema <henk at oegema.com> wrote:
        What is the meaning of getting (regular) this message:
        .....[WARNING] sofia_reg.c:1241 SIP auth challenge (REGISTER) on
        sofia profile 'internal' for [1001 at 10.0.0.101] from ip
        10.0.0.12 ??
        
        Extension 1001 is a Portech GSM gateway MV370. 
        
        
        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org
        


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From mitch.capper at gmail.com  Wed Jun 15 02:55:19 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Tue, 14 Jun 2011 15:55:19 -0700
Subject: [Freeswitch-users] Trying to get TLS/SRTP working
In-Reply-To: <BANLkTimLoVv7sRw-OOY-th0fC94uj-oYzw@mail.gmail.com>
References: <BANLkTimLoVv7sRw-OOY-th0fC94uj-oYzw@mail.gmail.com>
Message-ID: <BANLkTimBeyxTPZVAH9T8iki9o33ak0mp+A@mail.gmail.com>

Are you using latest trunk or what version of freeswitch?  Can you
post the rest of the startup log so we can see if it possibly failed
later?

~Mitch

On Tue, Jun 14, 2011 at 11:01 AM, Rob Hutton <justlikeef at gmail.com> wrote:
> Trying to get TLS working.? When sofia loads, I see that it seems to accept
> the parameters:
>
> http://pastebin.freeswitch.org/16487
>
> But sofia status shows nothing running on the TLS ports:
> ???????????????????? Name
> Type?????????????????????????????????????? Data????? State
> =================================================================================================
> ?????????? sipinterface_3?????? profile
> sip:mod_sofia at 192.168.2.25:5080????? RUNNING (0)
> ?????????? sipinterface_2?????? profile
> sip:mod_sofia at 192.168.2.25:5070????? RUNNING (0)
> ???????????? 192.168.2.25???????? alias
> sipinterface_1????? ALIASED
> ?????????? sipinterface_1?????? profile
> sip:mod_sofia at 192.168.2.25:5060????? RUNNING (0)
> ????????????? voicemail_1???????? alias
> sipinterface_1????? ALIASED
> =================================================================================================
> 3 profiles 2 aliases
>
>
> /usr/local/freeswitch/conf/ssl looks like:
> -rw-r--r-- 1 root root 3627 Jun 14 12:55 agent.pem
> -rw-r--r-- 1 root root 1996 Jun 14 13:04 cafile.pem
>
>
> Freeswitch is compiled with SSL support:
>
> ldd freeswitch
> ??????? linux-vdso.so.1 =>? (0x00007fff415ff000)
> ??????? libm.so.6 => /lib64/libm.so.6 (0x00007fab636b8000)
> ??????? libfreeswitch.so.1 => /usr/local/freeswitch/lib/libfreeswitch.so.1
> (0x00007fab632c1000)
> ??????? libuuid.so.1 => /lib64/libuuid.so.1 (0x00007fab630bc000)
> ??????? librt.so.1 => /lib64/librt.so.1 (0x00007fab62eb3000)
> ??????? libcrypt.so.1 => /lib64/libcrypt.so.1 (0x00007fab62c78000)
> ??????? libpthread.so.0 => /lib64/libpthread.so.0 (0x00007fab62a5b000)
> ??????? libssl.so.1.0.0 => /lib64/libssl.so.1.0.0 (0x00007fab627ff000)
> ??????? libcrypto.so.1.0.0 => /lib64/libcrypto.so.1.0.0 (0x00007fab6244e000)
> ??????? libdl.so.2 => /lib64/libdl.so.2 (0x00007fab6224a000)
> ??????? libz.so.1 => /lib64/libz.so.1 (0x00007fab62032000)
> ??????? libncurses.so.5 => /lib64/libncurses.so.5 (0x00007fab61ddd000)
> ??????? libc.so.6 => /lib64/libc.so.6 (0x00007fab61a70000)
> ??????? libstdc++.so.6 => /usr/lib64/libstdc++.so.6 (0x00007fab61767000)
> ??????? libgcc_s.so.1 => /lib64/libgcc_s.so.1 (0x00007fab61551000)
> ??????? libodbc.so.1 => /usr/lib64/libodbc.so.1 (0x00007fab612e5000)
> ??????? /lib64/ld-linux-x86-64.so.2 (0x00007fab6390f000)
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From brokendash at gmail.com  Wed Jun 15 02:56:50 2011
From: brokendash at gmail.com (broken dash)
Date: Tue, 14 Jun 2011 17:56:50 -0500
Subject: [Freeswitch-users] mod_pocketsphinx
In-Reply-To: <CE8866A5-FAB1-429E-A911-C3EF0625651C@freeswitch.org>
References: <4C1997A0.1080909@gmail.com>
	<BA0A1EAE-474B-4FE1-8BB3-03BC4B1C3655@freeswitch.org>
	<4C1ACD2A.70906@gmail.com>
	<D7122611-F34A-40B4-B0BE-C95A3D551E09@freeswitch.org>
	<4C1AE301.30405@gmail.com> <4C3D48A5.6030508@todandlorna.com>
	<C4BCDAB3-5371-48D5-810F-1A8F299529F6@freeswitch.org>
	<AANLkTi=aTwdANAkNR1V5HXnnrZD8FDeRVWKJktWYCGFa@mail.gmail.com>
	<AF932101-9B49-43F6-A488-8E1C5897E88F@freeswitch.org>
	<AANLkTimOskco=-=uDGdU8OPTOnrU0LXD7iAQyKFo8-4e@mail.gmail.com>
	<CE8866A5-FAB1-429E-A911-C3EF0625651C@freeswitch.org>
Message-ID: <BANLkTi=7uDtTx4xWJr9tTnwMNuEQvxZXEQ@mail.gmail.com>

Has this been fixed? Perhaps a workaround or something... It will
build on Debian Squeeze, but it damn sure doesn't work. :-)


freeswitch at 127.0.0.1@internal> version
FreeSWITCH Version 1.0.head (git-36f812d 2011-06-14 00-35-18 -0400)

freeswitch at 127.0.0.1@internal> load mod_pocketsphinx
+OK Reloading XML
-ERR [module load file routine returned an error]

2011-06-14 17:46:31.721075 [INFO] switch_time.c:1020 Timezone reloaded
530 definitions
freeswitch at 127.0.0.1@internal> 2011-06-14 17:46:31.721075 [CRIT]
switch_loadable_module.c:928 Error Loading module
/usr/local/freeswitch/mod/mod_pocketsphinx.so
**/usr/local/freeswitch/mod/mod_pocketsphinx.so: undefined symbol:
ngram_model_get_counts**



On Wed, Dec 8, 2010 at 9:16 AM, Brian West <brian at freeswitch.org> wrote:
> Compiles fine on CentOS not sure what your issue is but someone that cares
> about Ubuntu should probably figure it out and post patches if possible.
> /b
> On Dec 8, 2010, at 9:02 AM, Jan Kubr wrote:
>
> Were you able to reproduce the problem? Not sure where to look to get rid of
> this :(
> Jan
>
> On Mon, Dec 6, 2010 at 8:19 PM, Brian West?<brian at freeswitch.org>?wrote:
>>
>> No clue I'll try to compile it again today.
>> /b
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From spencer at 5ninesolutions.com  Wed Jun 15 02:58:30 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Tue, 14 Jun 2011 15:58:30 -0700
Subject: [Freeswitch-users] mod_cidlookup crashing
Message-ID: <0E43D250-4B7E-453F-91F2-E4253DCC1068@5ninesolutions.com>

Hello all,
When I enable caching in mod_cidlookup, a query causes a crash.  Is there some configuration error or any way I can troubleshoot this?  There isn't any output other than an immediate crash and a core dump.

memcache status verbose
Lib version: 0.32
Servers: 1
  localhost (11211)
    pid: 9598
    uptime: 5126
    time: 1308091953
    version: 1.4.5
    pointer_size: 64
    rusage_user: 0.71989
    rusage_system: 0.15997
    curr_items: 0
    total_items: 0
    bytes: 0
    curr_connections: 10
    total_connections: 32
    connection_structures: 11
    cmd_get: 18
    cmd_set: 0
    get_hits: 0
    get_misses: 18
    evictions: 0
    bytes_read: 0
    bytes_written: 0
    limit_maxbytes: 67108864
    threads: 4

freeswitch at internal> cidlookup status
+OK
 url: http://cnam.bulkcnam.com/?id=XXXX&did=${caller_id_number}
 cache: true
 cache-expire: 86400
 curl-timeout: 2000
 curl-warn-duration: 1000
 odbc-dsn: clidlookup_odbc
 sql: (null)
 citystate-sql:       SELECT CONCAT(ratecenter, ' ', state) as name       FROM npa_nxx_company_ocn       WHERE npa = ${caller_id_number:1:3} AND nxx = ${caller_id_number:4:3}       LIMIT 1       
 ODBC Compiled: true

Thanks,
Spencer







From cyril.zlachevsky at gmail.com  Wed Jun 15 02:13:27 2011
From: cyril.zlachevsky at gmail.com (Cyril Zlachevsky)
Date: Wed, 15 Jun 2011 01:13:27 +0300
Subject: [Freeswitch-users] ACL
In-Reply-To: <BANLkTimGGNr1u-rS3F4hyk2oYebHvz-6=A@mail.gmail.com>
References: <4DEEBF6D.7050608@gmail.com>
	<BANLkTimGGNr1u-rS3F4hyk2oYebHvz-6=A@mail.gmail.com>
Message-ID: <4DF7DD07.7080504@gmail.com>

Yes, you are right - I'm prefer flat file because customizing the default config is nightmare for me.
I changed "allow" to "deny" in my freeswitch.xml with absolutely no effect - anyone still register 
on my FS from any IP!
I read http://wiki.freeswitch.org/wiki/Acl but can't stand how enable ACL for my SIP UA's.


08.06.2011 10:51, Steven Ayre ?????:
> <list name="strict" *default="allow"*>
> <node type="allow" cidr="195.225.XXX.XXX/32"/>
> </list>
>
> You want a default of 'deny'. The default is what to do with any IP not on the list, what you've set
> allows every IP.
>
> I would suggest you build your config off the default config by the way, customising them to suit
> you and removing what you don't need. The directory structure is there to help you manage things
> (what you've posted makes it look like you might be using a flat file?). There's a lot of parameters
> you've missed out setting.
>
> For example:
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>
> These will have absolutely no effect. They set a global variable that you can use elsewhere in your
> configuration, but you're not actually using them. You should be setting the inbound & outbound
> codec preferences on the sofia profile in a param, either using the global variable e.g.
> $${outbound_codec_prefs} or by setting them explicitly there.
>
> -Steve
>
>
> On 8 June 2011 01:16, Cyril Zlachevsky <cyril.zlachevsky at gmail.com
> <mailto:cyril.zlachevsky at gmail.com>> wrote:
>
>     Hi,
>     Can't stand what should I do for ACL working.
>
>     I'm forwarding a call from a particular static IP to number at voipprovider.
>     I refused of using FS default configs because they are complicated and redundant for me.
>
>     This is my configuration:
>     <?xml version="1.0"?>
>     <document type="freeswitch/xml">
>
>     <X-PRE-PROCESS cmd="set" data="local_address=$${local_ip_v4}"/>
>
>     <X-PRE-PROCESS cmd="set" data="sipnet_proxy=voiprovider.com <http://voiprovider.com>"/>
>     <X-PRE-PROCESS cmd="set" data="sipnet_login=echo"/>
>     <X-PRE-PROCESS cmd="set" data="sipnet_password=echo"/>
>
>     <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/>
>
>     <section name="configuration">
>
>     <configuration name="switch.conf">
>     <settings>
>     <param name="colorize-console" value="true"/>
>     <param name="max-sessions" value="1000"/>
>     <param name="sessions-per-second" value="30"/>
>     <param name="loglevel" value="debug"/>
>     </settings>
>     </configuration>
>
>     <configuration name="modules.conf">
>     <modules>
>     <load module="mod_console"/>
>     <load module="mod_logfile"/>
>     <load module="mod_sofia"/>
>     <load module="mod_dialplan_xml"/>
>     <load module="mod_dptools"/>
>     <load module="mod_commands"/>
>     <load module="mod_expr"/>
>     <load module="mod_sndfile"/>
>     </modules>
>     </configuration>
>
>     <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
>     <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>
>     <configuration name="console.conf">
>     <mappings>
>     <map name="all" value="console,debug,info,notice,warning,err,crit,alert"/>
>     </mappings>
>     <settings>
>     <param name="colorize" value="true"/>
>     <param name="loglevel" value="info"/>
>     </settings>
>     </configuration>
>
>     <configuration name="logfile.conf">
>     <settings>
>     <param name="rotate-on-hup" value="true"/>
>     </settings>
>     <profiles>
>     <profile name="default">
>     <settings>
>     <param name="logfile" value="/var/log/freeswitch/freeswitch.log"/>
>     <param name="rollover" value="10485760"/>
>     </settings>
>     <mappings>
>     <map name="all" value="debug,info,notice,warning,err,crit,alert"/>
>     </mappings>
>     </profile>
>     </profiles>
>     </configuration>
>
>     <configuration name="acl.conf" description="Network Lists">
>     <network-lists>
>     <list name="strict" default="allow">
>     <node type="allow" cidr="195.225.XXX.XXX/32"/>
>     </list>
>     <list name="domains" default="deny">
>     <node type="allow" domain="$${domain}"/>
>     </list>
>     </network-lists>
>     </configuration>
>
>     <configuration name="sofia.conf">
>     <global_settings>
>     <param name="log-level" value="0"/>
>     <param name="debug-presence" value="0"/>
>     </global_settings>
>     <profiles>
>     <profile name="local">
>     <domains>
>     <domain name="all" alias="true" parse="false"/>
>     </domains>
>     <settings>
>     <param name="context" value="local"/>
>     <param name="sip-port" value="5060"/>
>     <param name="dialplan" value="XML"/>
>     <param name="rtp-ip" value="$${local_address}"/>
>     <param name="sip-ip" value="$${local_address}"/>
>     </settings>
>     </profile>
>     <profile name="sipnet">
>     <domains>
>     <domain name="all" alias="false" parse="false"/>
>     </domains>
>     <settings>
>     <param name="context" value="sipnet"/>
>     <param name="sip-port" value="5080"/>
>     <param name="dialplan" value="XML"/>
>     <param name="rtp-ip" value="$${local_address}"/>
>     <param name="sip-ip" value="$${local_address}"/>
>     <!--param name="auth-calls" value="false"/-->
>     <param name="auth-calls" value="true"/>
>     <param name="apply-inbound-acl" value="strict"/>
>     </settings>
>     <gateways>
>     <gateway name="sipnet">
>     <param name="proxy" value="$${sipnet_proxy}"/>
>     <param name="realm" value="asterisk"/>
>     <param name="username" value="$${sipnet_login}"/>
>     <param name="password" value="$${sipnet_password}"/>
>     <param name="local-network-acl" value="strict"/>
>     </gateway>
>     </gateways>
>     </profile>
>     </profiles>
>     </configuration>
>     </section>
>
>     <section name="dialplan">
>     <context name="local">
>     <extension name="local-accounts">
>     <condition field="destination_number" expression="^(10[1-3])$">
>     <action application="bridge" data="user/$1@$${local_address}"/>
>     </condition>
>     </extension>
>     <extension name="sipnet">
>     <condition field="destination_number" expression="^(\d+)$">
>     <action application="set" data="effective_caller_id_number=$${sipnet_login}"/>
>     <action application="bridge"
>     data="{sip_invite_domain=$${sipnet_proxy}}sofia/sipnet/$1@$${sipnet_proxy}"/>
>     </condition>
>     </extension>
>     </context>
>     </section>
>
>     <section name="directory">
>     <domain name="$${local_address}">
>     <params>
>     <param name="dial-string"
>     value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>     </params>
>     <groups>
>     <group name="local">
>     <users>
>     <user id="inboundtest"></user>
>     <params>
>     <param name="debug" value="yes"/>
>     <param name="register" value="true"/>
>     <param name="apply-inbound-acl" value="strict"/>
>     <param name="apply-register-acl" value="strict"/>
>     </params>
>     </users>
>     </group>
>     </groups>
>     </domain>
>     </section>
>     </document>
>
>     When I start FS, I can't see my IP 195.225.XXX.XXX in freeswitch.log - only this:
>     [NOTICE] switch_core.c:1088 Created ip list rfc1918.auto default (deny)
>     [NOTICE] switch_utils.c:248 Adding 10.0.0.0/8 <http://10.0.0.0/8> (allow) [] to list rfc1918.auto
>     [NOTICE] switch_utils.c:248 Adding 172.16.0.0/12 <http://172.16.0.0/12> (allow) [] to list
>     rfc1918.auto
>     [NOTICE] switch_utils.c:248 Adding 192.168.0.0/16 <http://192.168.0.0/16> (allow) [] to list
>     rfc1918.auto
>     [NOTICE] switch_core.c:1096 Created ip list wan.auto default (allow)
>     [NOTICE] switch_utils.c:248 Adding 10.0.0.0/8 <http://10.0.0.0/8> (deny) [] to list wan.auto
>     [NOTICE] switch_utils.c:248 Adding 172.16.0.0/12 <http://172.16.0.0/12> (deny) [] to list wan.auto
>     [NOTICE] switch_utils.c:248 Adding 192.168.0.0/16 <http://192.168.0.0/16> (deny) [] to list wan.auto
>     [NOTICE] switch_core.c:1104 Created ip list nat.auto default (deny)
>     [NOTICE] switch_core.c:1106 Adding 88.198.XXX.XXX/255.255.255.255 <http://255.255.255.255>
>     (deny) to list nat.auto
>     [NOTICE] switch_utils.c:248 Adding 10.0.0.0/8 <http://10.0.0.0/8> (allow) [] to list nat.auto
>     [NOTICE] switch_utils.c:248 Adding 172.16.0.0/12 <http://172.16.0.0/12> (allow) [] to list nat.auto
>     [NOTICE] switch_utils.c:248 Adding 192.168.0.0/16 <http://192.168.0.0/16> (allow) [] to list
>     nat.auto
>     [NOTICE] switch_core.c:1115 Created ip list loopback.auto default (deny)
>     [NOTICE] switch_utils.c:248 Adding 127.0.0.0/8 <http://127.0.0.0/8> (allow) [] to list loopback.auto
>     [NOTICE] switch_core.c:1121 Created ip list localnet.auto default (deny)
>     [NOTICE] switch_core.c:1124 Adding 88.198.XXX.XXX/255.255.255.255 <http://255.255.255.255>
>     (allow) to list localnet.auto
>
>     With my current configuration FS allow to register from any IP.
>     Where is my error?
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From justlikeef at gmail.com  Wed Jun 15 03:52:50 2011
From: justlikeef at gmail.com (Rob Hutton)
Date: Tue, 14 Jun 2011 19:52:50 -0400
Subject: [Freeswitch-users] Trying to get TLS/SRTP working
In-Reply-To: <BANLkTimBeyxTPZVAH9T8iki9o33ak0mp+A@mail.gmail.com>
References: <BANLkTimLoVv7sRw-OOY-th0fC94uj-oYzw@mail.gmail.com>
	<BANLkTimBeyxTPZVAH9T8iki9o33ak0mp+A@mail.gmail.com>
Message-ID: <201106141952.50449.justlikeef@gmail.com>

This is running a git version from mid day yesterday.

That is the complete (non-debug) log of a mod_sofia reload.  I can capture a 
log from a full load or turn logging up, but there isn't anything later from 
mod_sofia than this.

On Tuesday 14 June 2011 18:55:19 Mitch Capper wrote:
> Are you using latest trunk or what version of freeswitch?  Can you
> post the rest of the startup log so we can see if it possibly failed
> later?
> 
> ~Mitch
> 
> On Tue, Jun 14, 2011 at 11:01 AM, Rob Hutton <justlikeef at gmail.com> wrote:
> > Trying to get TLS working.  When sofia loads, I see that it seems to
> > accept the parameters:
> > 
> > http://pastebin.freeswitch.org/16487
> > 
> > But sofia status shows nothing running on the TLS ports:
> >                      Name
> > Type                                       Data      State
> > =========================================================================
> > ======================== sipinterface_3       profile
> > sip:mod_sofia at 192.168.2.25:5080      RUNNING (0)
> >            sipinterface_2       profile
> > sip:mod_sofia at 192.168.2.25:5070      RUNNING (0)
> >              192.168.2.25         alias
> > sipinterface_1      ALIASED
> >            sipinterface_1       profile
> > sip:mod_sofia at 192.168.2.25:5060      RUNNING (0)
> >               voicemail_1         alias
> > sipinterface_1      ALIASED
> > =========================================================================
> > ======================== 3 profiles 2 aliases
> > 
> > 
> > /usr/local/freeswitch/conf/ssl looks like:
> > -rw-r--r-- 1 root root 3627 Jun 14 12:55 agent.pem
> > -rw-r--r-- 1 root root 1996 Jun 14 13:04 cafile.pem
> > 
> > 
> > Freeswitch is compiled with SSL support:
> > 
> > ldd freeswitch
> >         linux-vdso.so.1 =>  (0x00007fff415ff000)
> >         libm.so.6 => /lib64/libm.so.6 (0x00007fab636b8000)
> >         libfreeswitch.so.1 =>
> > /usr/local/freeswitch/lib/libfreeswitch.so.1 (0x00007fab632c1000)
> >         libuuid.so.1 => /lib64/libuuid.so.1 (0x00007fab630bc000)
> >         librt.so.1 => /lib64/librt.so.1 (0x00007fab62eb3000)
> >         libcrypt.so.1 => /lib64/libcrypt.so.1 (0x00007fab62c78000)
> >         libpthread.so.0 => /lib64/libpthread.so.0 (0x00007fab62a5b000)
> >         libssl.so.1.0.0 => /lib64/libssl.so.1.0.0 (0x00007fab627ff000)
> >         libcrypto.so.1.0.0 => /lib64/libcrypto.so.1.0.0
> > (0x00007fab6244e000) libdl.so.2 => /lib64/libdl.so.2
> > (0x00007fab6224a000)
> >         libz.so.1 => /lib64/libz.so.1 (0x00007fab62032000)
> >         libncurses.so.5 => /lib64/libncurses.so.5 (0x00007fab61ddd000)
> >         libc.so.6 => /lib64/libc.so.6 (0x00007fab61a70000)
> >         libstdc++.so.6 => /usr/lib64/libstdc++.so.6 (0x00007fab61767000)
> >         libgcc_s.so.1 => /lib64/libgcc_s.so.1 (0x00007fab61551000)
> >         libodbc.so.1 => /usr/lib64/libodbc.so.1 (0x00007fab612e5000)
> >         /lib64/ld-linux-x86-64.so.2 (0x00007fab6390f000)
> > 
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From justlikeef at gmail.com  Wed Jun 15 04:07:04 2011
From: justlikeef at gmail.com (Rob Hutton)
Date: Tue, 14 Jun 2011 20:07:04 -0400
Subject: [Freeswitch-users] Trying to get TLS/SRTP working
In-Reply-To: <BANLkTimBeyxTPZVAH9T8iki9o33ak0mp+A@mail.gmail.com>
References: <BANLkTimLoVv7sRw-OOY-th0fC94uj-oYzw@mail.gmail.com>
	<BANLkTimBeyxTPZVAH9T8iki9o33ak0mp+A@mail.gmail.com>
Message-ID: <BANLkTim4jk=gEW6ZdusrZSXeA6uit--o1A@mail.gmail.com>

Full startup log:
http://pastebin.freeswitch.org/16494

Here's the config file:
http://pastebin.freeswitch.org/16495

FreeSWITCH version: 1.0.head (git-d52a254 2011-06-13 18-27-28 -0400)

On Tue, Jun 14, 2011 at 6:55 PM, Mitch Capper <mitch.capper at gmail.com>wrote:

> Are you using latest trunk or what version of freeswitch?  Can you
> post the rest of the startup log so we can see if it possibly failed
> later?
>
> ~Mitch
>
> On Tue, Jun 14, 2011 at 11:01 AM, Rob Hutton <justlikeef at gmail.com> wrote:
> > Trying to get TLS working.  When sofia loads, I see that it seems to
> accept
> > the parameters:
> >
> > http://pastebin.freeswitch.org/16487
> >
> > But sofia status shows nothing running on the TLS ports:
> >                      Name
> > Type                                       Data      State
> >
> =================================================================================================
> >            sipinterface_3       profile
> > sip:mod_sofia at 192.168.2.25:5080      RUNNING (0)
> >            sipinterface_2       profile
> > sip:mod_sofia at 192.168.2.25:5070      RUNNING (0)
> >              192.168.2.25         alias
> > sipinterface_1      ALIASED
> >            sipinterface_1       profile
> > sip:mod_sofia at 192.168.2.25:5060      RUNNING (0)
> >               voicemail_1         alias
> > sipinterface_1      ALIASED
> >
> =================================================================================================
> > 3 profiles 2 aliases
> >
> >
> > /usr/local/freeswitch/conf/ssl looks like:
> > -rw-r--r-- 1 root root 3627 Jun 14 12:55 agent.pem
> > -rw-r--r-- 1 root root 1996 Jun 14 13:04 cafile.pem
> >
> >
> > Freeswitch is compiled with SSL support:
> >
> > ldd freeswitch
> >         linux-vdso.so.1 =>  (0x00007fff415ff000)
> >         libm.so.6 => /lib64/libm.so.6 (0x00007fab636b8000)
> >         libfreeswitch.so.1 =>
> /usr/local/freeswitch/lib/libfreeswitch.so.1
> > (0x00007fab632c1000)
> >         libuuid.so.1 => /lib64/libuuid.so.1 (0x00007fab630bc000)
> >         librt.so.1 => /lib64/librt.so.1 (0x00007fab62eb3000)
> >         libcrypt.so.1 => /lib64/libcrypt.so.1 (0x00007fab62c78000)
> >         libpthread.so.0 => /lib64/libpthread.so.0 (0x00007fab62a5b000)
> >         libssl.so.1.0.0 => /lib64/libssl.so.1.0.0 (0x00007fab627ff000)
> >         libcrypto.so.1.0.0 => /lib64/libcrypto.so.1.0.0
> (0x00007fab6244e000)
> >         libdl.so.2 => /lib64/libdl.so.2 (0x00007fab6224a000)
> >         libz.so.1 => /lib64/libz.so.1 (0x00007fab62032000)
> >         libncurses.so.5 => /lib64/libncurses.so.5 (0x00007fab61ddd000)
> >         libc.so.6 => /lib64/libc.so.6 (0x00007fab61a70000)
> >         libstdc++.so.6 => /usr/lib64/libstdc++.so.6 (0x00007fab61767000)
> >         libgcc_s.so.1 => /lib64/libgcc_s.so.1 (0x00007fab61551000)
> >         libodbc.so.1 => /usr/lib64/libodbc.so.1 (0x00007fab612e5000)
> >         /lib64/ld-linux-x86-64.so.2 (0x00007fab6390f000)
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From qingquan at globalroam.com  Wed Jun 15 05:55:35 2011
From: qingquan at globalroam.com (qingquan luo)
Date: Wed, 15 Jun 2011 09:55:35 +0800
Subject: [Freeswitch-users] in bypass_media mode,
	2833 payload type modified by freeswitch!
Message-ID: <BANLkTi=HpPE7q4Kbrkwe-KJxj69SYJtx0Q@mail.gmail.com>

Hi All,

  I use freeswitch to bridge one incoming call to other target phone number.
  I use bypass_media mode.
  So the rtp is not go through the freeswitch,
  By I notice that When LegB reply 183 or 200 SDP message. Freeswitch
change it 2833 telephone-event payload type. and forwarding the message to
caller. This make the caller use wrong dtmf payload type to send dtmf

  What wrong with that?  How can fix it?
  Any information or help is welcome.

information with "sofia global siptrace on"
http://pastebin.freeswitch.org/16496

Thanks


Best Regards


Qingquan

-- 
Using Gmail? Please read this important notice:
http://www.fsf.org/campaigns/jstrap/gmail?40922.
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From brad at tech21.com  Wed Jun 15 07:30:46 2011
From: brad at tech21.com (Brad Mina)
Date: Tue, 14 Jun 2011 20:30:46 -0700
Subject: [Freeswitch-users] Trying to get TLS/SRTP working
In-Reply-To: <BANLkTim4jk=gEW6ZdusrZSXeA6uit--o1A@mail.gmail.com>
References: <BANLkTimLoVv7sRw-OOY-th0fC94uj-oYzw@mail.gmail.com>
	<BANLkTimBeyxTPZVAH9T8iki9o33ak0mp+A@mail.gmail.com>
	<BANLkTim4jk=gEW6ZdusrZSXeA6uit--o1A@mail.gmail.com>
Message-ID: <BANLkTi=VOZE=f4oxrb3HOY5V4t3DrVNa9A@mail.gmail.com>

Line 89 and 90 of your startup log clearly show TLS starting on port 5061 on
your internal profile.

2011-06-14 19:58:19.325605 [DEBUG] sofia.c:3100 tls-port [5061]
>
> 2011-06-14 19:58:19.325610 [DEBUG] sofia.c:3100 tls-version [tlsv1]
>
>
My 'sofia status' returns about the same thing as you:

> freeswitch at internal> sofia status
>
>                      Name   Type                                      Data
>> State
>
>
>> =================================================================================================
>
>                  internal profile         sip:mod_sofia at pbx.ip.add.ress
>> :5060 RUNNING (0)
>
>             internal-ipv6 profile                  sip:mod_sofia@
>> [::1]:5060 RUNNING (0)
>
>                  external profile         sip:mod_sofia at pbx.ip.add.ress
>> :5080 RUNNING (0)
>
>    external::bw-secondary gateway
>> sip:1831xxxxxxx at 216.82.225.202 NOREG
>
>         external::bw-main gateway
>> sip:1831xxxxxxx at 216.82.224.202 NOREG
>
>            pbx.ip.add.ress  alias
>>  internal ALIASED
>
>
>> =================================================================================================
>
> 3 profiles 1 alias
>
>
>
Have you tried configuring a phone for use with TLS? If so what problems are
you having with it?


On Tue, Jun 14, 2011 at 5:07 PM, Rob Hutton <justlikeef at gmail.com> wrote:

> Full startup log:
> http://pastebin.freeswitch.org/16494
>
> Here's the config file:
> http://pastebin.freeswitch.org/16495
>
> FreeSWITCH version: 1.0.head (git-d52a254 2011-06-13 18-27-28 -0400)
>
>
> On Tue, Jun 14, 2011 at 6:55 PM, Mitch Capper <mitch.capper at gmail.com>wrote:
>
>> Are you using latest trunk or what version of freeswitch?  Can you
>> post the rest of the startup log so we can see if it possibly failed
>> later?
>>
>> ~Mitch
>>
>> On Tue, Jun 14, 2011 at 11:01 AM, Rob Hutton <justlikeef at gmail.com>
>> wrote:
>> > Trying to get TLS working.  When sofia loads, I see that it seems to
>> accept
>> > the parameters:
>> >
>> > http://pastebin.freeswitch.org/16487
>> >
>> > But sofia status shows nothing running on the TLS ports:
>> >                      Name
>> > Type                                       Data      State
>> >
>> =================================================================================================
>> >            sipinterface_3       profile
>> > sip:mod_sofia at 192.168.2.25:5080      RUNNING (0)
>> >            sipinterface_2       profile
>> > sip:mod_sofia at 192.168.2.25:5070      RUNNING (0)
>> >              192.168.2.25         alias
>> > sipinterface_1      ALIASED
>> >            sipinterface_1       profile
>> > sip:mod_sofia at 192.168.2.25:5060      RUNNING (0)
>> >               voicemail_1         alias
>> > sipinterface_1      ALIASED
>> >
>> =================================================================================================
>> > 3 profiles 2 aliases
>> >
>> >
>> > /usr/local/freeswitch/conf/ssl looks like:
>> > -rw-r--r-- 1 root root 3627 Jun 14 12:55 agent.pem
>> > -rw-r--r-- 1 root root 1996 Jun 14 13:04 cafile.pem
>> >
>> >
>> > Freeswitch is compiled with SSL support:
>> >
>> > ldd freeswitch
>> >         linux-vdso.so.1 =>  (0x00007fff415ff000)
>> >         libm.so.6 => /lib64/libm.so.6 (0x00007fab636b8000)
>> >         libfreeswitch.so.1 =>
>> /usr/local/freeswitch/lib/libfreeswitch.so.1
>> > (0x00007fab632c1000)
>> >         libuuid.so.1 => /lib64/libuuid.so.1 (0x00007fab630bc000)
>> >         librt.so.1 => /lib64/librt.so.1 (0x00007fab62eb3000)
>> >         libcrypt.so.1 => /lib64/libcrypt.so.1 (0x00007fab62c78000)
>> >         libpthread.so.0 => /lib64/libpthread.so.0 (0x00007fab62a5b000)
>> >         libssl.so.1.0.0 => /lib64/libssl.so.1.0.0 (0x00007fab627ff000)
>> >         libcrypto.so.1.0.0 => /lib64/libcrypto.so.1.0.0
>> (0x00007fab6244e000)
>> >         libdl.so.2 => /lib64/libdl.so.2 (0x00007fab6224a000)
>> >         libz.so.1 => /lib64/libz.so.1 (0x00007fab62032000)
>> >         libncurses.so.5 => /lib64/libncurses.so.5 (0x00007fab61ddd000)
>> >         libc.so.6 => /lib64/libc.so.6 (0x00007fab61a70000)
>> >         libstdc++.so.6 => /usr/lib64/libstdc++.so.6 (0x00007fab61767000)
>> >         libgcc_s.so.1 => /lib64/libgcc_s.so.1 (0x00007fab61551000)
>> >         libodbc.so.1 => /usr/lib64/libodbc.so.1 (0x00007fab612e5000)
>> >         /lib64/ld-linux-x86-64.so.2 (0x00007fab6390f000)
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sid.kshatriya at gmail.com  Wed Jun 15 09:43:37 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Wed, 15 Jun 2011 11:13:37 +0530
Subject: [Freeswitch-users] ACL
In-Reply-To: <4DF7DD07.7080504@gmail.com>
References: <4DEEBF6D.7050608@gmail.com>
	<BANLkTimGGNr1u-rS3F4hyk2oYebHvz-6=A@mail.gmail.com>
	<4DF7DD07.7080504@gmail.com>
Message-ID: <BANLkTim-UBTQwK8ufY89ZcLPDARRXhjAZw@mail.gmail.com>

Have you tried typing

reloadacl reloadxml

from the freeswitch command line?


On Wed, Jun 15, 2011 at 3:43 AM, Cyril Zlachevsky <
cyril.zlachevsky at gmail.com> wrote:

> Yes, you are right - I'm prefer flat file because customizing the default
> config is nightmare for me.
> I changed "allow" to "deny" in my freeswitch.xml with absolutely no effect
> - anyone still register
> on my FS from any IP!
> I read http://wiki.freeswitch.org/wiki/Acl but can't stand how enable ACL
> for my SIP UA's.
>
>
> 08.06.2011 10:51, Steven Ayre ?????:
> > <list name="strict" *default="allow"*>
> > <node type="allow" cidr="195.225.XXX.XXX/32"/>
> > </list>
> >
> > You want a default of 'deny'. The default is what to do with any IP not
> on the list, what you've set
> > allows every IP.
> >
> > I would suggest you build your config off the default config by the way,
> customising them to suit
> > you and removing what you don't need. The directory structure is there to
> help you manage things
> > (what you've posted makes it look like you might be using a flat file?).
> There's a lot of parameters
> > you've missed out setting.
> >
> > For example:
> > <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h
> ,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
> > <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
> >
> > These will have absolutely no effect. They set a global variable that you
> can use elsewhere in your
> > configuration, but you're not actually using them. You should be setting
> the inbound & outbound
> > codec preferences on the sofia profile in a param, either using the
> global variable e.g.
> > $${outbound_codec_prefs} or by setting them explicitly there.
> >
> > -Steve
> >
> >
> > On 8 June 2011 01:16, Cyril Zlachevsky <cyril.zlachevsky at gmail.com
> > <mailto:cyril.zlachevsky at gmail.com>> wrote:
> >
> >     Hi,
> >     Can't stand what should I do for ACL working.
> >
> >     I'm forwarding a call from a particular static IP to
> number at voipprovider.
> >     I refused of using FS default configs because they are complicated
> and redundant for me.
> >
> >     This is my configuration:
> >     <?xml version="1.0"?>
> >     <document type="freeswitch/xml">
> >
> >     <X-PRE-PROCESS cmd="set" data="local_address=$${local_ip_v4}"/>
> >
> >     <X-PRE-PROCESS cmd="set" data="sipnet_proxy=voiprovider.com <
> http://voiprovider.com>"/>
> >     <X-PRE-PROCESS cmd="set" data="sipnet_login=echo"/>
> >     <X-PRE-PROCESS cmd="set" data="sipnet_password=echo"/>
> >
> >     <X-PRE-PROCESS cmd="set"
> data="sound_prefix=$${sounds_dir}/en/us/callie"/>
> >
> >     <section name="configuration">
> >
> >     <configuration name="switch.conf">
> >     <settings>
> >     <param name="colorize-console" value="true"/>
> >     <param name="max-sessions" value="1000"/>
> >     <param name="sessions-per-second" value="30"/>
> >     <param name="loglevel" value="debug"/>
> >     </settings>
> >     </configuration>
> >
> >     <configuration name="modules.conf">
> >     <modules>
> >     <load module="mod_console"/>
> >     <load module="mod_logfile"/>
> >     <load module="mod_sofia"/>
> >     <load module="mod_dialplan_xml"/>
> >     <load module="mod_dptools"/>
> >     <load module="mod_commands"/>
> >     <load module="mod_expr"/>
> >     <load module="mod_sndfile"/>
> >     </modules>
> >     </configuration>
> >
> >     <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h
> ,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
> >     <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
> >
> >     <configuration name="console.conf">
> >     <mappings>
> >     <map name="all"
> value="console,debug,info,notice,warning,err,crit,alert"/>
> >     </mappings>
> >     <settings>
> >     <param name="colorize" value="true"/>
> >     <param name="loglevel" value="info"/>
> >     </settings>
> >     </configuration>
> >
> >     <configuration name="logfile.conf">
> >     <settings>
> >     <param name="rotate-on-hup" value="true"/>
> >     </settings>
> >     <profiles>
> >     <profile name="default">
> >     <settings>
> >     <param name="logfile" value="/var/log/freeswitch/freeswitch.log"/>
> >     <param name="rollover" value="10485760"/>
> >     </settings>
> >     <mappings>
> >     <map name="all" value="debug,info,notice,warning,err,crit,alert"/>
> >     </mappings>
> >     </profile>
> >     </profiles>
> >     </configuration>
> >
> >     <configuration name="acl.conf" description="Network Lists">
> >     <network-lists>
> >     <list name="strict" default="allow">
> >     <node type="allow" cidr="195.225.XXX.XXX/32"/>
> >     </list>
> >     <list name="domains" default="deny">
> >     <node type="allow" domain="$${domain}"/>
> >     </list>
> >     </network-lists>
> >     </configuration>
> >
> >     <configuration name="sofia.conf">
> >     <global_settings>
> >     <param name="log-level" value="0"/>
> >     <param name="debug-presence" value="0"/>
> >     </global_settings>
> >     <profiles>
> >     <profile name="local">
> >     <domains>
> >     <domain name="all" alias="true" parse="false"/>
> >     </domains>
> >     <settings>
> >     <param name="context" value="local"/>
> >     <param name="sip-port" value="5060"/>
> >     <param name="dialplan" value="XML"/>
> >     <param name="rtp-ip" value="$${local_address}"/>
> >     <param name="sip-ip" value="$${local_address}"/>
> >     </settings>
> >     </profile>
> >     <profile name="sipnet">
> >     <domains>
> >     <domain name="all" alias="false" parse="false"/>
> >     </domains>
> >     <settings>
> >     <param name="context" value="sipnet"/>
> >     <param name="sip-port" value="5080"/>
> >     <param name="dialplan" value="XML"/>
> >     <param name="rtp-ip" value="$${local_address}"/>
> >     <param name="sip-ip" value="$${local_address}"/>
> >     <!--param name="auth-calls" value="false"/-->
> >     <param name="auth-calls" value="true"/>
> >     <param name="apply-inbound-acl" value="strict"/>
> >     </settings>
> >     <gateways>
> >     <gateway name="sipnet">
> >     <param name="proxy" value="$${sipnet_proxy}"/>
> >     <param name="realm" value="asterisk"/>
> >     <param name="username" value="$${sipnet_login}"/>
> >     <param name="password" value="$${sipnet_password}"/>
> >     <param name="local-network-acl" value="strict"/>
> >     </gateway>
> >     </gateways>
> >     </profile>
> >     </profiles>
> >     </configuration>
> >     </section>
> >
> >     <section name="dialplan">
> >     <context name="local">
> >     <extension name="local-accounts">
> >     <condition field="destination_number" expression="^(10[1-3])$">
> >     <action application="bridge" data="user/$1@$${local_address}"/>
> >     </condition>
> >     </extension>
> >     <extension name="sipnet">
> >     <condition field="destination_number" expression="^(\d+)$">
> >     <action application="set"
> data="effective_caller_id_number=$${sipnet_login}"/>
> >     <action application="bridge"
> >     data="{sip_invite_domain=$${sipnet_proxy}}sofia/sipnet/$1@
> $${sipnet_proxy}"/>
> >     </condition>
> >     </extension>
> >     </context>
> >     </section>
> >
> >     <section name="directory">
> >     <domain name="$${local_address}">
> >     <params>
> >     <param name="dial-string"
> >     value="{presence_id=${dialed_user}@
> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
> >     </params>
> >     <groups>
> >     <group name="local">
> >     <users>
> >     <user id="inboundtest"></user>
> >     <params>
> >     <param name="debug" value="yes"/>
> >     <param name="register" value="true"/>
> >     <param name="apply-inbound-acl" value="strict"/>
> >     <param name="apply-register-acl" value="strict"/>
> >     </params>
> >     </users>
> >     </group>
> >     </groups>
> >     </domain>
> >     </section>
> >     </document>
> >
> >     When I start FS, I can't see my IP 195.225.XXX.XXX in freeswitch.log
> - only this:
> >     [NOTICE] switch_core.c:1088 Created ip list rfc1918.auto default
> (deny)
> >     [NOTICE] switch_utils.c:248 Adding 10.0.0.0/8 <http://10.0.0.0/8>
> (allow) [] to list rfc1918.auto
> >     [NOTICE] switch_utils.c:248 Adding 172.16.0.0/12 <
> http://172.16.0.0/12> (allow) [] to list
> >     rfc1918.auto
> >     [NOTICE] switch_utils.c:248 Adding 192.168.0.0/16 <
> http://192.168.0.0/16> (allow) [] to list
> >     rfc1918.auto
> >     [NOTICE] switch_core.c:1096 Created ip list wan.auto default (allow)
> >     [NOTICE] switch_utils.c:248 Adding 10.0.0.0/8 <http://10.0.0.0/8>
> (deny) [] to list wan.auto
> >     [NOTICE] switch_utils.c:248 Adding 172.16.0.0/12 <
> http://172.16.0.0/12> (deny) [] to list wan.auto
> >     [NOTICE] switch_utils.c:248 Adding 192.168.0.0/16 <
> http://192.168.0.0/16> (deny) [] to list wan.auto
> >     [NOTICE] switch_core.c:1104 Created ip list nat.auto default (deny)
> >     [NOTICE] switch_core.c:1106 Adding 88.198.XXX.XXX/255.255.255.255 <
> http://255.255.255.255>
> >     (deny) to list nat.auto
> >     [NOTICE] switch_utils.c:248 Adding 10.0.0.0/8 <http://10.0.0.0/8>
> (allow) [] to list nat.auto
> >     [NOTICE] switch_utils.c:248 Adding 172.16.0.0/12 <
> http://172.16.0.0/12> (allow) [] to list nat.auto
> >     [NOTICE] switch_utils.c:248 Adding 192.168.0.0/16 <
> http://192.168.0.0/16> (allow) [] to list
> >     nat.auto
> >     [NOTICE] switch_core.c:1115 Created ip list loopback.auto default
> (deny)
> >     [NOTICE] switch_utils.c:248 Adding 127.0.0.0/8 <http://127.0.0.0/8>
> (allow) [] to list loopback.auto
> >     [NOTICE] switch_core.c:1121 Created ip list localnet.auto default
> (deny)
> >     [NOTICE] switch_core.c:1124 Adding 88.198.XXX.XXX/255.255.255.255 <
> http://255.255.255.255>
> >     (allow) to list localnet.auto
> >
> >     With my current configuration FS allow to register from any IP.
> >     Where is my error?
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org <mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sidharth Kshatriya
www.sidk.info
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From avi at avimarcus.net  Wed Jun 15 10:26:46 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 15 Jun 2011 09:26:46 +0300
Subject: [Freeswitch-users] [Fwd: Re: Warning - SIP auth challenge]
In-Reply-To: <1308091686.2247.16.camel@DELL>
References: <1308091686.2247.16.camel@DELL>
Message-ID: <BANLkTinL14GqQ32+aqkLTFJCkE0hDsMp+Q@mail.gmail.com>

If it's from the same extension and the same device, then it's set to
register every 4 minutes. If you don't want that, you can change the
register time on the phone or forces the subscription expire time:
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#force-subscription-expires

-Avi Marcus


On Wed, Jun 15, 2011 at 1:48 AM, Henk Oegema <henk at oegema.com> wrote:

>  But the message is every 4 minutes.
>
> -------- Forwarded Message --------
> *From*: Avi Marcus <avi at avimarcus.net<Avi%20Marcus%20%3cavi at avimarcus.net%3e>
> >
> *Reply-to*: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *To*: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<FreeSWITCH%20Users%20Help%20%3cfreeswitch-users at lists.freeswitch.org%3e>
> >
> *Subject*: Re: [Freeswitch-users] Warning - SIP auth challenge
> *Date*: Wed, 15 Jun 2011 01:38:29 +0300
>
>
> It's just a notice that there's a register attempt. If it's every 30 or 60
> minutes from a known extension, it's fine.
>
> It's there for fail2ban to notice excessive register attempts so it can
> block abuse from clients that try to register but don't submit a password to
> avoid setting off the warning bells.
>
> -Avi
>
>
>
>
> On Wed, Jun 15, 2011 at 1:33 AM, Henk Oegema <henk at oegema.com> wrote:
>
> What is the meaning of getting (regular) this message:
> .....[WARNING] sofia_reg.c:1241 SIP auth challenge (REGISTER) on sofia
> profile 'internal' for [1001 at 10.0.0.101] from ip 10.0.0.12 ??
>
> Extension 1001 is a Portech GSM gateway MV370.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Wed Jun 15 11:15:09 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 15 Jun 2011 09:15:09 +0200
Subject: [Freeswitch-users] in bypass_media mode,
	2833 payload type modified by freeswitch!
In-Reply-To: <BANLkTi=HpPE7q4Kbrkwe-KJxj69SYJtx0Q@mail.gmail.com>
References: <BANLkTi=HpPE7q4Kbrkwe-KJxj69SYJtx0Q@mail.gmail.com>
Message-ID: <4A5154F6-90DE-4B40-9F24-E3E0D0824D02@ipeva.fr>

You have a mail issue. Check your spam...

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

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

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




Le 15/06/2011 ? 03:55, qingquan luo a ?crit :

> Hi All,
> 
>   I use freeswitch to bridge one incoming call to other target phone number.
>   I use bypass_media mode.
>   So the rtp is not go through the freeswitch,
>   By I notice that When LegB reply 183 or 200 SDP message. Freeswitch
> change it 2833 telephone-event payload type. and forwarding the message to
> caller. This make the caller use wrong dtmf payload type to send dtmf
> 
>   What wrong with that?  How can fix it?
>   Any information or help is welcome.
> 
> information with "sofia global siptrace on"
> http://pastebin.freeswitch.org/16496
> 
> Thanks
> 
> 
> Best Regards
> 
> 
> Qingquan
> 
> -- 
> Using Gmail? Please read this important notice: http://www.fsf.org/campaigns/jstrap/gmail?40922.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From sid.kshatriya at gmail.com  Wed Jun 15 11:48:11 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Wed, 15 Jun 2011 13:18:11 +0530
Subject: [Freeswitch-users] Want to play music on hold while executing a
 time consuming operation in Lua
In-Reply-To: <BANLkTinXFZPeprzatd45N42ZuP2obvU5JQ@mail.gmail.com>
References: <BANLkTinXFZPeprzatd45N42ZuP2obvU5JQ@mail.gmail.com>
Message-ID: <BANLkTi=LwrT+0reVP=K7wONGujDT28RWgA@mail.gmail.com>

Any help guys ?

On Tue, Jun 14, 2011 at 8:51 PM, Sidharth Kshatriya <sid.kshatriya at gmail.com
> wrote:

> I am implementing an IVR using Lua in Freeswitch. In my Lua script I use
> curl to a web service. Sometimes the response takes a long time to come
> back. During that time period I would like to play music on hold.
>
> I have searched the freeswitch discussion archives a lot. There seem to be
> many suggested ways to implement music on hold from a Lua script but the
> answers are not very clear / not really applicable to my use case. I don't
> know what method I should use and which one is recommended.
>
> *Method 1:* Transfer the call to 9664 (music on hold extension). However
> the implementation for this solution for this does not seem to be available
> in Lua. For example: How would I transfer the call back?
> *Method 2:* Using bgapi uuid_park park the call and using uuid_broadcast
> play an audio file. Again what do I do to unpark the call..?
>
> Thanks,
>
> Sidharth
> --
> Sidharth Kshatriya
> www.sidk.info
>
>


-- 
Sidharth Kshatriya
www.sidk.info
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From ankitwalia4u at gmail.com  Wed Jun 15 12:34:36 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Wed, 15 Jun 2011 14:04:36 +0530
Subject: [Freeswitch-users] Invalid Profile [external]
In-Reply-To: <BANLkTinWjFsoFi-o526EY6G=fDLgobdOXw@mail.gmail.com>
References: <BANLkTinMcV_hY0yrg-FeD+4Aag+u+oB_=w@mail.gmail.com>
	<BANLkTimqAa=4+m6ni6Exac5xX9fU8eSiKA@mail.gmail.com>
	<BANLkTi=BxWtAbiBGnfAtu+x2U2qhiGZDbQ@mail.gmail.com>
	<BANLkTinAjO4YOt9y5vMR1q0c0s244EBD8A@mail.gmail.com>
	<BANLkTin+kELEx-ZCr8+Scq4RRu6e6RKofg@mail.gmail.com>
	<BANLkTinWjFsoFi-o526EY6G=fDLgobdOXw@mail.gmail.com>
Message-ID: <BANLkTi=ZxQBQ_j8ierfVYXqAgy4-NHQNxg@mail.gmail.com>

Thanks MC and Christian. I was doing a silly thing, I did not realize. I
made FreeSwitch as Windows Service, so I was starting FS as service and as
well as from Debug->FS. I checked in the logs and then I find that same logs
created two times. it triggered my mind that there could be problem because
of it. Then, I started FS from Debug->FS. Now the profile is loaded and GW
is also registered.

I thinkthe problem is when i started FS as Win Service. It used my port
5080. Now, when I am starting it next time with Debug->FS application. it
could not get the port free and failed to load. Am I right?

But, I have one question:
1. In real scenario, Do we need to use FS as Windows Service or to start the
FS as exe. What if we start FS as Service and we would like to do some test
using commands as FS CLI?

If, the above question/explanation sounding weird to you, sorry for that. I
am very new to FreeSwitch.

Regards,
Ankit

On Tue, Jun 14, 2011 at 8:48 PM, Michael Collins <msc at freeswitch.org> wrote:

> Be sure that the external profile is actually coming up. Watch the log when
> you start FS - if the external profile's port is already in use then the
> profile will not start.
> -MC
>
>
> On Tue, Jun 14, 2011 at 8:05 AM, ankIT WALiA <ankitwalia4u at gmail.com>wrote:
>
>> Where I can modify the path for the default configuration for the external
>> sip-profile is
>> /usr/local/freeswitch/conf/
>> sip_profiles/external.xml(Modify the path
>> according to your Win7-installation).
>>
>> I have seen, everywhere there is a relative path.
>>
>>
>> On Tue, Jun 14, 2011 at 8:26 PM, ankIT WALiA <ankitwalia4u at gmail.com>wrote:
>>
>>> Christian,
>>>
>>> I tried commenting out some code one after another, if there is any typo.
>>> I got the error Invalid profile every time. I even tried removing
>>> iptel.xml and add the content in external.xml, still issue persist.
>>>
>>> Even for default configuration, this error seems to be coming.
>>>
>>> Thanks
>>> Ankit
>>>
>>>
>>> On Tue, Jun 14, 2011 at 6:40 PM, Christian Benke <benkokakao at gmail.com>wrote:
>>>
>>>> On 14 June 2011 14:51, Christian Benke <benkokakao at gmail.com> wrote:
>>>> >> Now, after launching FS CLI, I run a "sofia profile external restart
>>>> >> reloadxml" which is giving me message "Invalid Profile [external]"
>>>> >
>>>> > Please give me the full output of the command above, i believe it says
>>>> > more than just "Invalid Profile".
>>>>
>>>> Oh! It's indeed just one line saying "2011-06-14 13:07:47.335914
>>>> [WARNING] sofia.c:4044 No Such Profile 'external'"
>>>>
>>>> You probably have a typo in your sip-profile-configuration, in case
>>>> you didn't change it intentionally.
>>>>
>>>> The default configuration for the external sip-profile is
>>>> /usr/local/freeswitch/conf/sip_profiles/external.xml(Modify the path
>>>> according to your Win7-installation).
>>>>
>>>> Please verify you have a section in your configuration that says:
>>>>
>>>> <profile name="external">
>>>>  <gateways>
>>>>    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
>>>>  </gateways>
>>>>
>>>> ...lots of other stuff in default-config...
>>>>
>>>> </profile>
>>>>
>>>>
>>>> Then you add the iptel-configuration you've posted above in
>>>> /usr/local/freeswitch/conf/sip_profiles/external/iptel.xml(If this
>>>> didn't happen already)
>>>>
>>>> Best regards,
>>>> Christian
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Life is like a rose its upto u feel it as its fragrance or thorns
>>>
>>
>>
>>
>> --
>> Life is like a rose its upto u feel it as its fragrance or thorns
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Life is like a rose its upto u feel it as its fragrance or thorns
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From steveayre at gmail.com  Wed Jun 15 13:12:42 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 15 Jun 2011 10:12:42 +0100
Subject: [Freeswitch-users] Invalid Profile [external]
In-Reply-To: <BANLkTi=ZxQBQ_j8ierfVYXqAgy4-NHQNxg@mail.gmail.com>
References: <BANLkTinMcV_hY0yrg-FeD+4Aag+u+oB_=w@mail.gmail.com>
	<BANLkTimqAa=4+m6ni6Exac5xX9fU8eSiKA@mail.gmail.com>
	<BANLkTi=BxWtAbiBGnfAtu+x2U2qhiGZDbQ@mail.gmail.com>
	<BANLkTinAjO4YOt9y5vMR1q0c0s244EBD8A@mail.gmail.com>
	<BANLkTin+kELEx-ZCr8+Scq4RRu6e6RKofg@mail.gmail.com>
	<BANLkTinWjFsoFi-o526EY6G=fDLgobdOXw@mail.gmail.com>
	<BANLkTi=ZxQBQ_j8ierfVYXqAgy4-NHQNxg@mail.gmail.com>
Message-ID: <BANLkTi=9mHYKMSmaa7dhscYMWZNARXksVg@mail.gmail.com>

>
> 1. In real scenario, Do we need to use FS as Windows Service or to start
> the FS as exe. What if we start FS as Service and we would like to do some
> test using commands as FS CLI?
>

There are 2 different consoles. Starting FS with the -c option gives you a
console provided by FS itself. I don't use FS on Windows but I assume that's
what Debug->FS was doing.

There's also the ESL protocol (provided by mod_event_socket) which allows
you to connect over the network, either from on the same server or from
another. That provides remote console access, and fs_cli will let you
connect to that.

-Steve



On 15 June 2011 09:34, ankIT WALiA <ankitwalia4u at gmail.com> wrote:
>
> Thanks MC and Christian. I was doing a silly thing, I did not realize. I
made FreeSwitch as Windows Service, so I was starting FS as service and as
well as from Debug->FS. I checked in the logs and then I find that same logs
created two times. it triggered my mind that there could be problem because
of it. Then, I started FS from Debug->FS. Now the profile is loaded and GW
is also registered.
> I thinkthe problem is when i started FS as Win Service. It used my port
5080. Now, when I am starting it next time with Debug->FS application. it
could not get the port free and failed to load. Am I right?
>
> But, I have one question:
> 1. In real scenario, Do we need to use FS as Windows Service or to start
the FS as exe. What if we start FS as Service and we would like to do some
test using commands as FS CLI?
> If, the above question/explanation sounding weird to you, sorry for that.
I am very new to FreeSwitch.
> Regards,
> Ankit
> On Tue, Jun 14, 2011 at 8:48 PM, Michael Collins <msc at freeswitch.org>
wrote:
>>
>> Be sure that the external profile is actually coming up. Watch the log
when you start FS - if the external profile's port is already in use then
the profile will not start.
>> -MC
>>
>> On Tue, Jun 14, 2011 at 8:05 AM, ankIT WALiA <ankitwalia4u at gmail.com>
wrote:
>>>
>>> Where I can modify the path for the default configuration for the
external sip-profile is
>>> /usr/local/freeswitch/conf/
>>> sip_profiles/external.xml(Modify the path
>>> according to your Win7-installation).
>>>
>>> I have seen, everywhere there is a relative path.
>>>
>>>
>>> On Tue, Jun 14, 2011 at 8:26 PM, ankIT WALiA <ankitwalia4u at gmail.com>
wrote:
>>>>
>>>> Christian,
>>>>
>>>> I tried commenting out some code one after another, if there is any
typo.
>>>> I got the error Invalid profile every time. I even tried removing
iptel.xml and add the content in external.xml, still issue persist.
>>>>
>>>> Even for default configuration, this error seems to be coming.
>>>>
>>>> Thanks
>>>> Ankit
>>>>
>>>> On Tue, Jun 14, 2011 at 6:40 PM, Christian Benke <benkokakao at gmail.com>
wrote:
>>>>>
>>>>> On 14 June 2011 14:51, Christian Benke <benkokakao at gmail.com> wrote:
>>>>> >> Now, after launching FS CLI, I run a "sofia profile external
restart
>>>>> >> reloadxml" which is giving me message "Invalid Profile [external]"
>>>>> >
>>>>> > Please give me the full output of the command above, i believe it
says
>>>>> > more than just "Invalid Profile".
>>>>>
>>>>> Oh! It's indeed just one line saying "2011-06-14 13:07:47.335914
>>>>> [WARNING] sofia.c:4044 No Such Profile 'external'"
>>>>>
>>>>> You probably have a typo in your sip-profile-configuration, in case
>>>>> you didn't change it intentionally.
>>>>>
>>>>> The default configuration for the external sip-profile is
>>>>> /usr/local/freeswitch/conf/sip_profiles/external.xml(Modify the path
>>>>> according to your Win7-installation).
>>>>>
>>>>> Please verify you have a section in your configuration that says:
>>>>>
>>>>> <profile name="external">
>>>>>  <gateways>
>>>>>    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
>>>>>  </gateways>
>>>>>
>>>>> ...lots of other stuff in default-config...
>>>>>
>>>>> </profile>
>>>>>
>>>>>
>>>>> Then you add the iptel-configuration you've posted above in
>>>>> /usr/local/freeswitch/conf/sip_profiles/external/iptel.xml(If this
>>>>> didn't happen already)
>>>>>
>>>>> Best regards,
>>>>> Christian
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> --
>>>> Life is like a rose its upto u feel it as its fragrance or thorns
>>>
>>>
>>>
>>> --
>>> Life is like a rose its upto u feel it as its fragrance or thorns
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Life is like a rose its upto u feel it as its fragrance or thorns
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From william at xofap.com  Wed Jun 15 13:18:24 2011
From: william at xofap.com (William Alianto)
Date: Wed, 15 Jun 2011 16:18:24 +0700
Subject: [Freeswitch-users] Asterisk as gateway for FS
Message-ID: <4DF878E0.80601@xofap.com>

Hi,

I'm trying to setup a system using Asterisk and FreeSwitch, where 
Asterisk will perform as gateway for FreeSwitch. Is this setup possible? 
If yes, where can I find an example setup to look on to?

-- 
Regards,

William



From benkokakao at gmail.com  Wed Jun 15 13:31:36 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Wed, 15 Jun 2011 11:31:36 +0200
Subject: [Freeswitch-users] Asterisk as gateway for FS
In-Reply-To: <4DF878E0.80601@xofap.com>
References: <4DF878E0.80601@xofap.com>
Message-ID: <BANLkTimS0hh-0MmjaiQb_Rs_3tQz_t2sUA@mail.gmail.com>

On 15 June 2011 11:18, William Alianto <william at xofap.com> wrote:
> I'm trying to setup a system using Asterisk and FreeSwitch, where
> Asterisk will perform as gateway for FreeSwitch. Is this setup possible?
> If yes, where can I find an example setup to look on to?

http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk

Regards,
Christian


From steveayre at gmail.com  Wed Jun 15 14:15:11 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 15 Jun 2011 11:15:11 +0100
Subject: [Freeswitch-users] [Fwd: Re: Warning - SIP auth challenge]
In-Reply-To: <BANLkTinL14GqQ32+aqkLTFJCkE0hDsMp+Q@mail.gmail.com>
References: <1308091686.2247.16.camel@DELL>
	<BANLkTinL14GqQ32+aqkLTFJCkE0hDsMp+Q@mail.gmail.com>
Message-ID: <BANLkTi=eaagMB6xuAskn_PUACo3i9Xt0=A@mail.gmail.com>

I think the log-auth-failures option will also disable logging of that
message, useful if you're not using fail2ban:
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#log-auth-failures

The only reason that it's a warning level message is so that you can run FS
on warning level and fail2ban can still see those messages.

-Steve


On 15 June 2011 07:26, Avi Marcus <avi at avimarcus.net> wrote:

> If it's from the same extension and the same device, then it's set to
> register every 4 minutes. If you don't want that, you can change the
> register time on the phone or forces the subscription expire time:
>
> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#force-subscription-expires
>
> -Avi Marcus
>
>
> On Wed, Jun 15, 2011 at 1:48 AM, Henk Oegema <henk at oegema.com> wrote:
>
>>  But the message is every 4 minutes.
>>
>> -------- Forwarded Message --------
>> *From*: Avi Marcus <avi at avimarcus.net<Avi%20Marcus%20%3cavi at avimarcus.net%3e>
>> >
>> *Reply-to*: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *To*: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<FreeSWITCH%20Users%20Help%20%3cfreeswitch-users at lists.freeswitch.org%3e>
>> >
>> *Subject*: Re: [Freeswitch-users] Warning - SIP auth challenge
>> *Date*: Wed, 15 Jun 2011 01:38:29 +0300
>>
>>
>> It's just a notice that there's a register attempt. If it's every 30 or 60
>> minutes from a known extension, it's fine.
>>
>> It's there for fail2ban to notice excessive register attempts so it can
>> block abuse from clients that try to register but don't submit a password to
>> avoid setting off the warning bells.
>>
>> -Avi
>>
>>
>>
>>
>> On Wed, Jun 15, 2011 at 1:33 AM, Henk Oegema <henk at oegema.com> wrote:
>>
>> What is the meaning of getting (regular) this message:
>> .....[WARNING] sofia_reg.c:1241 SIP auth challenge (REGISTER) on sofia
>> profile 'internal' for [1001 at 10.0.0.101] from ip 10.0.0.12 ??
>>
>> Extension 1001 is a Portech GSM gateway MV370.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From henk at oegema.com  Wed Jun 15 14:37:45 2011
From: henk at oegema.com (Henk Oegema)
Date: Wed, 15 Jun 2011 12:37:45 +0200
Subject: [Freeswitch-users] [Fwd:  Asterisk as gateway for FS]
Message-ID: <1308134265.2150.1.camel@DELL>

Just to be curios: What is the purpose of Asterisk being a gateway for
FS ?

Rgds.
Henk

-------- Forwarded Message --------
From: William Alianto <william at xofap.com>
Reply-to: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Asterisk as gateway for FS
Date: Wed, 15 Jun 2011 16:18:24 +0700


Hi,

I'm trying to setup a system using Asterisk and FreeSwitch, where 
Asterisk will perform as gateway for FreeSwitch. Is this setup possible? 
If yes, where can I find an example setup to look on to?

-- 
Regards,

William


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From fdelawarde at wirelessmundi.com  Wed Jun 15 16:09:45 2011
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Wed, 15 Jun 2011 14:09:45 +0200
Subject: [Freeswitch-users] custom ODBC timeouts
Message-ID: <1308139785.16213.140.camel@luna.madrid.commsmundi.com>

Hello everyone,

I would like to manually set the timeouts for ODBC (login, query and
connections), to use with DBH in lua, in order to quickly fallback to
another action if something goes wrong.

Those timeouts look like they are hard-coded in FS, but is there any way
to force them externally (odbc.ini or similar)?

Thanks,
Fran?ois.



From fdelawarde at wirelessmundi.com  Wed Jun 15 16:41:19 2011
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Wed, 15 Jun 2011 14:41:19 +0200
Subject: [Freeswitch-users] [Fwd:  Asterisk as gateway for FS]
In-Reply-To: <1308134265.2150.1.camel@DELL>
References: <1308134265.2150.1.camel@DELL>
Message-ID: <1308141679.16213.159.camel@luna.madrid.commsmundi.com>

It is possible and works very well, just connect some SIP trunk between
the two.

Here we use it for TDM, IAX, and chan_mobile call termination.

Regards,
Fran?ois.

On Wed, 2011-06-15 at 12:37 +0200, Henk Oegema wrote:
> Just to be curios: What is the purpose of Asterisk being a gateway for
> FS ?
> 
> Rgds.
> Henk
> 
> -------- Forwarded Message --------
> From: William Alianto <william at xofap.com>
> Reply-to: FreeSWITCH Users Help
> <freeswitch-users at lists.freeswitch.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] Asterisk as gateway for FS
> Date: Wed, 15 Jun 2011 16:18:24 +0700
> 
> Hi,
> 
> I'm trying to setup a system using Asterisk and FreeSwitch, where 
> Asterisk will perform as gateway for FreeSwitch. Is this setup possible? 
> If yes, where can I find an example setup to look on to?
> 
> -- 
> Regards,
> 
> William
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From justlikeef at gmail.com  Wed Jun 15 16:59:21 2011
From: justlikeef at gmail.com (Rob Hutton)
Date: Wed, 15 Jun 2011 08:59:21 -0400
Subject: [Freeswitch-users] Trying to get TLS/SRTP working
In-Reply-To: <BANLkTi=VOZE=f4oxrb3HOY5V4t3DrVNa9A@mail.gmail.com>
References: <BANLkTimLoVv7sRw-OOY-th0fC94uj-oYzw@mail.gmail.com>
	<BANLkTim4jk=gEW6ZdusrZSXeA6uit--o1A@mail.gmail.com>
	<BANLkTi=VOZE=f4oxrb3HOY5V4t3DrVNa9A@mail.gmail.com>
Message-ID: <201106150859.22504.justlikeef@gmail.com>

If I force TLS on the phone, it will not register.

A netstat shows nothing listening on the ports.

On Tuesday 14 June 2011 23:30:46 Brad Mina wrote:
> Line 89 and 90 of your startup log clearly show TLS starting on port 5061
> on your internal profile.
> 
> 2011-06-14 19:58:19.325605 [DEBUG] sofia.c:3100 tls-port [5061]
> 
> > 2011-06-14 19:58:19.325610 [DEBUG] sofia.c:3100 tls-version [tlsv1]
> 
> My 'sofia status' returns about the same thing as you:
> > freeswitch at internal> sofia status
> > 
> >                      Name   Type                                     
> >                      Data
> >> 
> >> State
> >> 
> >> 
> >> ========================================================================
> >> =========================
> >> 
> >                  internal profile         sip:mod_sofia at pbx.ip.add.ress
> >> :
> >> :5060 RUNNING (0)
> >> :
> >             internal-ipv6 profile                  sip:mod_sofia@
> >> 
> >> [::1]:5060 RUNNING (0)
> >> 
> >                  external profile         sip:mod_sofia at pbx.ip.add.ress
> >> :
> >> :5080 RUNNING (0)
> >> :
> >    external::bw-secondary gateway
> >> 
> >> sip:1831xxxxxxx at 216.82.225.202 NOREG
> >> 
> >         external::bw-main gateway
> >> 
> >> sip:1831xxxxxxx at 216.82.224.202 NOREG
> >> 
> >            pbx.ip.add.ress  alias
> >>  
> >>  internal ALIASED
> >> 
> >> ========================================================================
> >> =========================
> > 
> > 3 profiles 1 alias
> 
> Have you tried configuring a phone for use with TLS? If so what problems
> are you having with it?
> 
> On Tue, Jun 14, 2011 at 5:07 PM, Rob Hutton <justlikeef at gmail.com> wrote:
> > Full startup log:
> > http://pastebin.freeswitch.org/16494
> > 
> > Here's the config file:
> > http://pastebin.freeswitch.org/16495
> > 
> > FreeSWITCH version: 1.0.head (git-d52a254 2011-06-13 18-27-28 -0400)
> > 
> > On Tue, Jun 14, 2011 at 6:55 PM, Mitch Capper 
<mitch.capper at gmail.com>wrote:
> >> Are you using latest trunk or what version of freeswitch?  Can you
> >> post the rest of the startup log so we can see if it possibly failed
> >> later?
> >> 
> >> ~Mitch
> >> 
> >> On Tue, Jun 14, 2011 at 11:01 AM, Rob Hutton <justlikeef at gmail.com>
> >> 
> >> wrote:
> >> > Trying to get TLS working.  When sofia loads, I see that it seems to
> >> 
> >> accept
> >> 
> >> > the parameters:
> >> > 
> >> > http://pastebin.freeswitch.org/16487
> >> > 
> >> > But sofia status shows nothing running on the TLS ports:
> >> >                      Name
> >> > 
> >> > Type                                       Data      State
> >> 
> >> ========================================================================
> >> =========================
> >> 
> >> >            sipinterface_3       profile
> >> > 
> >> > sip:mod_sofia at 192.168.2.25:5080      RUNNING (0)
> >> > 
> >> >            sipinterface_2       profile
> >> > 
> >> > sip:mod_sofia at 192.168.2.25:5070      RUNNING (0)
> >> > 
> >> >              192.168.2.25         alias
> >> > 
> >> > sipinterface_1      ALIASED
> >> > 
> >> >            sipinterface_1       profile
> >> > 
> >> > sip:mod_sofia at 192.168.2.25:5060      RUNNING (0)
> >> > 
> >> >               voicemail_1         alias
> >> > 
> >> > sipinterface_1      ALIASED
> >> 
> >> ========================================================================
> >> =========================
> >> 
> >> > 3 profiles 2 aliases
> >> > 
> >> > 
> >> > /usr/local/freeswitch/conf/ssl looks like:
> >> > -rw-r--r-- 1 root root 3627 Jun 14 12:55 agent.pem
> >> > -rw-r--r-- 1 root root 1996 Jun 14 13:04 cafile.pem
> >> > 
> >> > 
> >> > Freeswitch is compiled with SSL support:
> >> > 
> >> > ldd freeswitch
> >> > 
> >> >         linux-vdso.so.1 =>  (0x00007fff415ff000)
> >> >         libm.so.6 => /lib64/libm.so.6 (0x00007fab636b8000)
> >> >         libfreeswitch.so.1 =>
> >> 
> >> /usr/local/freeswitch/lib/libfreeswitch.so.1
> >> 
> >> > (0x00007fab632c1000)
> >> > 
> >> >         libuuid.so.1 => /lib64/libuuid.so.1 (0x00007fab630bc000)
> >> >         librt.so.1 => /lib64/librt.so.1 (0x00007fab62eb3000)
> >> >         libcrypt.so.1 => /lib64/libcrypt.so.1 (0x00007fab62c78000)
> >> >         libpthread.so.0 => /lib64/libpthread.so.0 (0x00007fab62a5b000)
> >> >         libssl.so.1.0.0 => /lib64/libssl.so.1.0.0 (0x00007fab627ff000)
> >> >         libcrypto.so.1.0.0 => /lib64/libcrypto.so.1.0.0
> >> 
> >> (0x00007fab6244e000)
> >> 
> >> >         libdl.so.2 => /lib64/libdl.so.2 (0x00007fab6224a000)
> >> >         libz.so.1 => /lib64/libz.so.1 (0x00007fab62032000)
> >> >         libncurses.so.5 => /lib64/libncurses.so.5 (0x00007fab61ddd000)
> >> >         libc.so.6 => /lib64/libc.so.6 (0x00007fab61a70000)
> >> >         libstdc++.so.6 => /usr/lib64/libstdc++.so.6
> >> >         (0x00007fab61767000) libgcc_s.so.1 => /lib64/libgcc_s.so.1
> >> >         (0x00007fab61551000) libodbc.so.1 => /usr/lib64/libodbc.so.1
> >> >         (0x00007fab612e5000) /lib64/ld-linux-x86-64.so.2
> >> >         (0x00007fab6390f000)
> >> > 
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> 
> >> > UNSUBSCRIBE:
> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> 
> >> > http://www.freeswitch.org
> >> 
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org


From potxoka at gmail.com  Wed Jun 15 17:38:14 2011
From: potxoka at gmail.com (Antonio)
Date: Wed, 15 Jun 2011 15:38:14 +0200
Subject: [Freeswitch-users] Proxy traffic security
In-Reply-To: <BANLkTi=MGq5LSn7j+UmDFQLR2p0RHSoPGQ@mail.gmail.com>
References: <4DEFF3FA.5050500@gmail.com>	<BANLkTincK9zEprZjuMRMo5c6vejuX8_22w@mail.gmail.com>	<4DF1E456.1030607@gmail.com>
	<BANLkTi=MGq5LSn7j+UmDFQLR2p0RHSoPGQ@mail.gmail.com>
Message-ID: <4DF8B5C6.2080104@gmail.com>

Thanks ! ;-)

El 10/06/11 11:46, Steven Ayre escribi?:
> Iptables will allow you to block all but your proxies from sending. It 
> should work (slightly) faster than the FS ACL because it'll block it 
> earlier in the packet processing.
>
> -Steve
>
>
> On 10 June 2011 10:31, Antonio <potxoka at gmail.com 
> <mailto:potxoka at gmail.com>> wrote:
>
>     Hi,
>
>     If I had thought also use iptables, but it was to know if exists
>     something better to put the ip's (to prevent IP spoofing). Thanks.
>
>     Greetings
>     Anto
>
>     El 09/06/11 16:01, Christian Benke escribi?:
>     > ACLs definitely would not hurt. Also think about setting up a simple
>     > firewall(e.g. Shorewall or just plain IPTables) to limit the
>     access to
>     > the ip's which MUST have access(e.g. SIP-Provider IPs), you can
>     never
>     > be sure if a software(like FreeSWITCH) doesn't have a bug that could
>     > be exploited.
>     >
>     > Cheers,
>     > Christian
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From wes-fs at 499x.com  Wed Jun 15 18:00:30 2011
From: wes-fs at 499x.com (Wes)
Date: Wed, 15 Jun 2011 09:00:30 -0500
Subject: [Freeswitch-users] invalid application lua
In-Reply-To: <BANLkTim01AOjuyx8jgCuaF4ki_khnr0UUg@mail.gmail.com>
References: <4DF7D71B.3020103@499x.com>
	<BANLkTim01AOjuyx8jgCuaF4ki_khnr0UUg@mail.gmail.com>
Message-ID: <4DF8BAFE.1000704@499x.com>

freeswitch at transcription-desktop> reload mod_lua
Error including /opt/freeswitch/conf/lang/de/*.xml (No such file or 
directory)
2011-06-15 09:00:17.969278 [ERR] switch_xml.c:1329 Couldnt open 
/opt/freeswitch/conf/lang/de/*.xml (No such file or directory)
Error including /opt/freeswitch/conf/lang/fr/*.xml (No such file or 
directory)
2011-06-15 09:00:18.008426 [ERR] switch_xml.c:1329 Couldnt open 
/opt/freeswitch/conf/lang/fr/*.xml (No such file or directory)
Error including /opt/freeswitch/conf/lang/ru/*.xml (No such file or 
directory)
2011-06-15 09:00:18.008426 [ERR] switch_xml.c:1329 Couldnt open 
/opt/freeswitch/conf/lang/ru/*.xml (No such file or directory)
2011-06-15 09:00:18.021491 [INFO] mod_enum.c:808 ENUM Reloaded
2011-06-15 09:00:18.021491 [INFO] switch_time.c:950 Timezone reloaded 
530 definitions
2011-06-15 09:00:18.022839 [CRIT] switch_loadable_module.c:926 Error 
Loading module /opt/freeswitch/mod/mod_lua.so
**/opt/freeswitch/mod/mod_lua.so: cannot open shared object file: No 
such file or directory**

-ERR unloading module [No such module!]
+OK Reloading XML
-ERR loading module [module load file routine returned an error]

On 6/14/2011 4:55 PM, Avi Marcus wrote:
> What happens when you run "reload mod_lua" from the CLI?
>
> -Avi Marcus
>
>
>
> On Wed, Jun 15, 2011 at 12:48 AM, Wes <wes-fs at 499x.com 
> <mailto:wes-fs at 499x.com>> wrote:
>
>     I'm getting "invalid application lua" however, in modules.conf,
>     the line
>     for <load module="mod_lua"/> is active and not commented out.
>
>     Any help is appreciated. Thanks.
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From infos at madovsky.org  Wed Jun 15 18:21:12 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 15 Jun 2011 10:21:12 -0400
Subject: [Freeswitch-users] [Fwd: Re: Warning - SIP auth challenge]
References: <1308091686.2247.16.camel@DELL><BANLkTinL14GqQ32+aqkLTFJCkE0hDsMp+Q@mail.gmail.com>
	<BANLkTi=eaagMB6xuAskn_PUACo3i9Xt0=A@mail.gmail.com>
Message-ID: <26DA9D570679447EBFF6F49A2657976C@e1705>

useful if you use fail2ban.... ;)
  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, June 15, 2011 6:15 AM
  Subject: Re: [Freeswitch-users] [Fwd: Re: Warning - SIP auth challenge]


  I think the log-auth-failures option will also disable logging of that
  message, useful if you're not using fail2ban:
  http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#log-auth-failures

  The only reason that it's a warning level message is so that you can run FS
  on warning level and fail2ban can still see those messages.

  -Steve



  On 15 June 2011 07:26, Avi Marcus <avi at avimarcus.net> wrote:

    If it's from the same extension and the same device, then it's set to register every 4 minutes. If you don't want that, you can change the register time on the phone or forces the subscription expire time:
    http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#force-subscription-expires


    -Avi Marcus




    On Wed, Jun 15, 2011 at 1:48 AM, Henk Oegema <henk at oegema.com> wrote:

      But the message is every 4 minutes.

      -------- Forwarded Message --------
      From: Avi Marcus <avi at avimarcus.net>
      Reply-to: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
      To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
      Subject: Re: [Freeswitch-users] Warning - SIP auth challenge
      Date: Wed, 15 Jun 2011 01:38:29 +0300


      It's just a notice that there's a register attempt. If it's every 30 or 60 minutes from a known extension, it's fine. 

      It's there for fail2ban to notice excessive register attempts so it can block abuse from clients that try to register but don't submit a password to avoid setting off the warning bells.

      -Avi




      On Wed, Jun 15, 2011 at 1:33 AM, Henk Oegema <henk at oegema.com> wrote: 
        What is the meaning of getting (regular) this message:
        .....[WARNING] sofia_reg.c:1241 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1001 at 10.0.0.101] from ip 10.0.0.12 ??

        Extension 1001 is a Portech GSM gateway MV370. 

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





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

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





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





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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From david.ponzone at ipeva.fr  Wed Jun 15 18:24:33 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 15 Jun 2011 16:24:33 +0200
Subject: [Freeswitch-users] invalid application lua
In-Reply-To: <4DF8BAFE.1000704@499x.com>
References: <4DF7D71B.3020103@499x.com>
	<BANLkTim01AOjuyx8jgCuaF4ki_khnr0UUg@mail.gmail.com>
	<4DF8BAFE.1000704@499x.com>
Message-ID: <876A820B-B0A3-4FDA-9F02-C55E8A261AAA@ipeva.fr>

Wes,

in FS source tree, edit modules.conf and enable mod_lua.
Then recompile.

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

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

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




Le 15/06/2011 ? 16:00, Wes a ?crit :

> freeswitch at transcription-desktop> reload mod_lua
> Error including /opt/freeswitch/conf/lang/de/*.xml (No such file or directory)
> 2011-06-15 09:00:17.969278 [ERR] switch_xml.c:1329 Couldnt open /opt/freeswitch/conf/lang/de/*.xml (No such file or directory)
> Error including /opt/freeswitch/conf/lang/fr/*.xml (No such file or directory)
> 2011-06-15 09:00:18.008426 [ERR] switch_xml.c:1329 Couldnt open /opt/freeswitch/conf/lang/fr/*.xml (No such file or directory)
> Error including /opt/freeswitch/conf/lang/ru/*.xml (No such file or directory)
> 2011-06-15 09:00:18.008426 [ERR] switch_xml.c:1329 Couldnt open /opt/freeswitch/conf/lang/ru/*.xml (No such file or directory)
> 2011-06-15 09:00:18.021491 [INFO] mod_enum.c:808 ENUM Reloaded
> 2011-06-15 09:00:18.021491 [INFO] switch_time.c:950 Timezone reloaded 530 definitions
> 2011-06-15 09:00:18.022839 [CRIT] switch_loadable_module.c:926 Error Loading module /opt/freeswitch/mod/mod_lua.so
> **/opt/freeswitch/mod/mod_lua.so: cannot open shared object file: No such file or directory**
> 
> -ERR unloading module [No such module!]
> +OK Reloading XML
> -ERR loading module [module load file routine returned an error]
> 
> On 6/14/2011 4:55 PM, Avi Marcus wrote:
>> 
>> What happens when you run "reload mod_lua" from the CLI?
>> 
>> -Avi Marcus
>> 
>> 
>> 
>> On Wed, Jun 15, 2011 at 12:48 AM, Wes <wes-fs at 499x.com> wrote:
>> I'm getting "invalid application lua" however, in modules.conf, the line
>> for <load module="mod_lua"/> is active and not commented out.
>> 
>> Any help is appreciated. Thanks.
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jaybinks at gmail.com  Wed Jun 15 18:29:36 2011
From: jaybinks at gmail.com (jay binks)
Date: Thu, 16 Jun 2011 00:29:36 +1000
Subject: [Freeswitch-users] [Fwd: Re: Warning - SIP auth challenge]
In-Reply-To: <26DA9D570679447EBFF6F49A2657976C@e1705>
References: <1308091686.2247.16.camel@DELL>
	<BANLkTinL14GqQ32+aqkLTFJCkE0hDsMp+Q@mail.gmail.com>
	<BANLkTi=eaagMB6xuAskn_PUACo3i9Xt0=A@mail.gmail.com>
	<26DA9D570679447EBFF6F49A2657976C@e1705>
Message-ID: <BANLkTiksPFzR1NhR6jhca1dUWTs5jJpw5g@mail.gmail.com>

and everyone SHOULD use fail2ban ... or your asking for trouble..
not many good reasons to avoid it .

J

On Thu, Jun 16, 2011 at 12:21 AM, Madovsky <infos at madovsky.org> wrote:

>  useful if you use fail2ban.... ;)
>
> ----- Original Message -----
> *From:* Steven Ayre <steveayre at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, June 15, 2011 6:15 AM
> *Subject:* Re: [Freeswitch-users] [Fwd: Re: Warning - SIP auth challenge]
>
> I think the log-auth-failures option will also disable logging of that
> message, useful if you're not using fail2ban:
> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#log-auth-failures
>
> The only reason that it's a warning level message is so that you can run FS
> on warning level and fail2ban can still see those messages.
>
> -Steve
>
>
> On 15 June 2011 07:26, Avi Marcus <avi at avimarcus.net> wrote:
>
>> If it's from the same extension and the same device, then it's set to
>> register every 4 minutes. If you don't want that, you can change the
>> register time on the phone or forces the subscription expire time:
>>
>> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#force-subscription-expires
>>
>>   -Avi Marcus
>>
>>
>> On Wed, Jun 15, 2011 at 1:48 AM, Henk Oegema <henk at oegema.com> wrote:
>>
>>> But the message is every 4 minutes.
>>>
>>> -------- Forwarded Message --------
>>> *From*: Avi Marcus <avi at avimarcus.net<Avi%20Marcus%20%3cavi at avimarcus.net%3e>
>>> >
>>> *Reply-to*: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org
>>> >
>>> *To*: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<FreeSWITCH%20Users%20Help%20%3cfreeswitch-users at lists.freeswitch.org%3e>
>>> >
>>> *Subject*: Re: [Freeswitch-users] Warning - SIP auth challenge
>>> *Date*: Wed, 15 Jun 2011 01:38:29 +0300
>>>
>>>
>>> It's just a notice that there's a register attempt. If it's every 30 or
>>> 60 minutes from a known extension, it's fine.
>>>
>>> It's there for fail2ban to notice excessive register attempts so it can
>>> block abuse from clients that try to register but don't submit a password to
>>> avoid setting off the warning bells.
>>>
>>> -Avi
>>>
>>>
>>>
>>>
>>> On Wed, Jun 15, 2011 at 1:33 AM, Henk Oegema <henk at oegema.com> wrote:
>>>
>>> What is the meaning of getting (regular) this message:
>>> .....[WARNING] sofia_reg.c:1241 SIP auth challenge (REGISTER) on sofia
>>> profile 'internal' for [1001 at 10.0.0.101] from ip 10.0.0.12 ??
>>>
>>> Extension 1001 is a Portech GSM gateway MV370.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sincerely

Jay
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From wes-fs at 499x.com  Wed Jun 15 19:14:09 2011
From: wes-fs at 499x.com (Wes)
Date: Wed, 15 Jun 2011 10:14:09 -0500
Subject: [Freeswitch-users] invalid application lua
In-Reply-To: <876A820B-B0A3-4FDA-9F02-C55E8A261AAA@ipeva.fr>
References: <4DF7D71B.3020103@499x.com>	<BANLkTim01AOjuyx8jgCuaF4ki_khnr0UUg@mail.gmail.com>	<4DF8BAFE.1000704@499x.com>
	<876A820B-B0A3-4FDA-9F02-C55E8A261AAA@ipeva.fr>
Message-ID: <4DF8CC41.4080102@499x.com>

I'm using the ubuntu packages, so I can't do exactly that, but I did 
find that:

sudo apt-get install freeswitch-lua

did the trick!

I'm sure I'll have more questions now when I start to try the lua 
scripting, but for now, I've got an extension hitting the hello.lua 
example script and it works fine...thanks for the help!

On 6/15/2011 9:24 AM, David Ponzone wrote:
> Wes,
>
> in FS source tree, edit modules.conf and enable mod_lua.
> Then recompile.
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service ClientIPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
> diffusion non autoris?e est interdite. Tout message ?lectronique est 
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au 
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
> n'?tes pas destinataire de ce message, merci de le d?truire 
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 15/06/2011 ? 16:00, Wes a ?crit :
>
>> freeswitch at transcription-desktop> reload mod_lua
>> Error including /opt/freeswitch/conf/lang/de/*.xml (No such file or 
>> directory)
>> 2011-06-15 09:00:17.969278 [ERR] switch_xml.c:1329 Couldnt open 
>> /opt/freeswitch/conf/lang/de/*.xml (No such file or directory)
>> Error including /opt/freeswitch/conf/lang/fr/*.xml (No such file or 
>> directory)
>> 2011-06-15 09:00:18.008426 [ERR] switch_xml.c:1329 Couldnt open 
>> /opt/freeswitch/conf/lang/fr/*.xml (No such file or directory)
>> Error including /opt/freeswitch/conf/lang/ru/*.xml (No such file or 
>> directory)
>> 2011-06-15 09:00:18.008426 [ERR] switch_xml.c:1329 Couldnt open 
>> /opt/freeswitch/conf/lang/ru/*.xml (No such file or directory)
>> 2011-06-15 09:00:18.021491 [INFO] mod_enum.c:808 ENUM Reloaded
>> 2011-06-15 09:00:18.021491 [INFO] switch_time.c:950 Timezone reloaded 
>> 530 definitions
>> 2011-06-15 09:00:18.022839 [CRIT] switch_loadable_module.c:926 Error 
>> Loading module /opt/freeswitch/mod/mod_lua.so
>> **/opt/freeswitch/mod/mod_lua.so: cannot open shared object file: No 
>> such file or directory**
>>
>> -ERR unloading module [No such module!]
>> +OK Reloading XML
>> -ERR loading module [module load file routine returned an error]
>>
>> On 6/14/2011 4:55 PM, Avi Marcus wrote:
>>> What happens when you run "reload mod_lua" from the CLI?
>>>
>>> -Avi Marcus
>>>
>>>
>>> On Wed, Jun 15, 2011 at 12:48 AM, Wes <wes-fs at 499x.com 
>>> <mailto:wes-fs at 499x.com>> wrote:
>>>
>>>     I'm getting "invalid application lua" however, in modules.conf,
>>>     the line
>>>     for <load module="mod_lua"/> is active and not commented out.
>>>
>>>     Any help is appreciated. Thanks.
>>>
>>>     _______________________________________________
>>>     FreeSWITCH-users mailing list
>>>     FreeSWITCH-users at lists.freeswitch.org
>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org 
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From kris at kriskinc.com  Wed Jun 15 19:18:00 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 15 Jun 2011 11:18:00 -0400
Subject: [Freeswitch-users] in bypass_media mode,
 2833 payload type modified by freeswitch!
In-Reply-To: <BANLkTi=HpPE7q4Kbrkwe-KJxj69SYJtx0Q@mail.gmail.com>
References: <BANLkTi=HpPE7q4Kbrkwe-KJxj69SYJtx0Q@mail.gmail.com>
Message-ID: <BANLkTinGkG0rLro5vUPugv6eX=M77_nygA@mail.gmail.com>

Certainly looks like a bug.  Make sure you're running latest and open
an issue on JIRA.

On Tue, Jun 14, 2011 at 9:55 PM, qingquan luo <qingquan at globalroam.com> wrote:
> Hi All,
> ? I use freeswitch to bridge one incoming call to other target phone number.
> ? I use bypass_media mode.
> ? So the rtp is not go through the freeswitch,
> ? By I notice that When LegB reply 183 or 200 SDP message. Freeswitch
> change it 2833 telephone-event payload type. and forwarding the message to
> caller. This make the caller use wrong dtmf payload type to send dtmf
> ? What wrong with that? ?How can fix it?
> ? Any information or help is welcome.
> information with "sofia global siptrace on"
> http://pastebin.freeswitch.org/16496
> Thanks
>
> Best Regards
>
> Qingquan
> --
> Using Gmail? Please read this important notice:
> http://www.fsf.org/campaigns/jstrap/gmail?40922.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner


From avi at avimarcus.net  Wed Jun 15 19:21:11 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 15 Jun 2011 18:21:11 +0300
Subject: [Freeswitch-users] invalid application lua
In-Reply-To: <4DF8CC41.4080102@499x.com>
References: <4DF7D71B.3020103@499x.com>
	<BANLkTim01AOjuyx8jgCuaF4ki_khnr0UUg@mail.gmail.com>
	<4DF8BAFE.1000704@499x.com>
	<876A820B-B0A3-4FDA-9F02-C55E8A261AAA@ipeva.fr>
	<4DF8CC41.4080102@499x.com>
Message-ID: <BANLkTi=+kmhb1VmWbvNObb7s42M__jFVag@mail.gmail.com>

Compile from git, compile from git, compile from git!
http://wiki.freeswitch.org/wiki/Installation_Guide
The one in the main ubuntu 10.04 repo is Version:
1.0.head~git.master.20101015.1-1 - that's 8 months old...


 -Avi Marcus

On Wed, Jun 15, 2011 at 6:14 PM, Wes <wes-fs at 499x.com> wrote:

>  I'm using the ubuntu packages, so I can't do exactly that, but I did find
> that:
>
> sudo apt-get install freeswitch-lua
>
> did the trick!
>
> I'm sure I'll have more questions now when I start to try the lua
> scripting, but for now, I've got an extension hitting the hello.lua example
> script and it works fine...thanks for the help!
>
> On 6/15/2011 9:24 AM, David Ponzone wrote:
>
> Wes,
>
>  in FS source tree, edit modules.conf and enable mod_lua.
> Then recompile.
>
>   David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
>  Service Client IPeva
>  tel:      0811 46 26 26
>  www.ipeva.fr  -   www.ipeva-studio.com
>
>  *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
>  Le 15/06/2011 ? 16:00, Wes a ?crit :
>
>  freeswitch at transcription-desktop> reload mod_lua
> Error including /opt/freeswitch/conf/lang/de/*.xml (No such file or
> directory)
> 2011-06-15 09:00:17.969278 [ERR] switch_xml.c:1329 Couldnt open
> /opt/freeswitch/conf/lang/de/*.xml (No such file or directory)
> Error including /opt/freeswitch/conf/lang/fr/*.xml (No such file or
> directory)
> 2011-06-15 09:00:18.008426 [ERR] switch_xml.c:1329 Couldnt open
> /opt/freeswitch/conf/lang/fr/*.xml (No such file or directory)
> Error including /opt/freeswitch/conf/lang/ru/*.xml (No such file or
> directory)
> 2011-06-15 09:00:18.008426 [ERR] switch_xml.c:1329 Couldnt open
> /opt/freeswitch/conf/lang/ru/*.xml (No such file or directory)
> 2011-06-15 09:00:18.021491 [INFO] mod_enum.c:808 ENUM Reloaded
> 2011-06-15 09:00:18.021491 [INFO] switch_time.c:950 Timezone reloaded 530
> definitions
> 2011-06-15 09:00:18.022839 [CRIT] switch_loadable_module.c:926 Error
> Loading module /opt/freeswitch/mod/mod_lua.so
> **/opt/freeswitch/mod/mod_lua.so: cannot open shared object file: No such
> file or directory**
>
> -ERR unloading module [No such module!]
> +OK Reloading XML
> -ERR loading module [module load file routine returned an error]
>
> On 6/14/2011 4:55 PM, Avi Marcus wrote:
>
> What happens when you run "reload mod_lua" from the CLI?
>
>   -Avi Marcus
>
>
> On Wed, Jun 15, 2011 at 12:48 AM, Wes <wes-fs at 499x.com> wrote:
>
>> I'm getting "invalid application lua" however, in modules.conf, the line
>> for <load module="mod_lua"/> is active and not commented out.
>>
>> Any help is appreciated. Thanks.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Jun 15 20:25:00 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 15 Jun 2011 09:25:00 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <BANLkTik6=bG7kPQRYLCz+=wgtNnX3Xr6nQ@mail.gmail.com>

Hello all!

Today's agenda is here:
http://wiki.freeswitch.org/wiki/FS_weekly_2011_06_15

We have a light agenda but we do want to talk about a few things, in
particular the FreeSWITCH Cookbook and the new mod_rtmp that was released on
Monday.

Talk to you soon,
Michael
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From wes-fs at 499x.com  Wed Jun 15 20:31:37 2011
From: wes-fs at 499x.com (Wes)
Date: Wed, 15 Jun 2011 11:31:37 -0500
Subject: [Freeswitch-users] invalid application lua
In-Reply-To: <BANLkTi=+kmhb1VmWbvNObb7s42M__jFVag@mail.gmail.com>
References: <4DF7D71B.3020103@499x.com>	<BANLkTim01AOjuyx8jgCuaF4ki_khnr0UUg@mail.gmail.com>	<4DF8BAFE.1000704@499x.com>	<876A820B-B0A3-4FDA-9F02-C55E8A261AAA@ipeva.fr>	<4DF8CC41.4080102@499x.com>
	<BANLkTi=+kmhb1VmWbvNObb7s42M__jFVag@mail.gmail.com>
Message-ID: <4DF8DE69.4000306@499x.com>

ok, I'll give it a try.  A coworker who's an Ubuntu-head set me up this way!

Can I just do this in a separate directory without messing with the 
current installation just to try it out?

On 6/15/2011 10:21 AM, Avi Marcus wrote:
> Compile from git, compile from git, compile from git! 
> http://wiki.freeswitch.org/wiki/Installation_Guide
> The one in the main ubuntu 10.04 repo is Version: 
> 1.0.head~git.master.20101015.1-1 - that's 8 months old...
>
>
> -Avi Marcus
>
>
> On Wed, Jun 15, 2011 at 6:14 PM, Wes <wes-fs at 499x.com 
> <mailto:wes-fs at 499x.com>> wrote:
>
>     I'm using the ubuntu packages, so I can't do exactly that, but I
>     did find that:
>
>     sudo apt-get install freeswitch-lua
>
>     did the trick!
>
>     I'm sure I'll have more questions now when I start to try the lua
>     scripting, but for now, I've got an extension hitting the
>     hello.lua example script and it works fine...thanks for the help!
>
>     On 6/15/2011 9:24 AM, David Ponzone wrote:
>>     Wes,
>>
>>     in FS source tree, edit modules.conf and enable mod_lua.
>>     Then recompile.
>>
>>     David Ponzone Direction Technique
>>     email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>     tel:      01 74 03 18 97
>>     gsm:   06 66 98 76 34
>>
>>     Service ClientIPeva
>>     tel:      0811 46 26 26
>>     www.ipeva.fr  - www.ipeva-studio.com
>>
>>     /Ce message et toutes les pi?ces jointes sont confidentiels et
>>     ?tablis ? l'intention exclusive de ses destinataires. Toute
>>     utilisation ou diffusion non autoris?e est interdite. Tout
>>     message ?lectronique est susceptible d'alt?ration.
>>     /*/IPeva/*/ d?cline toute responsabilit? au titre de ce message
>>     s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>     destinataire de ce message, merci de le d?truire imm?diatement et
>>     d'avertir l'exp?diteur./
>>     /
>>     /
>>
>>
>>
>>     Le 15/06/2011 ? 16:00, Wes a ?crit :
>>
>>>     freeswitch at transcription-desktop> reload mod_lua
>>>     Error including /opt/freeswitch/conf/lang/de/*.xml (No such file
>>>     or directory)
>>>     2011-06-15 09:00:17.969278 [ERR] switch_xml.c:1329 Couldnt open
>>>     /opt/freeswitch/conf/lang/de/*.xml (No such file or directory)
>>>     Error including /opt/freeswitch/conf/lang/fr/*.xml (No such file
>>>     or directory)
>>>     2011-06-15 09:00:18.008426 [ERR] switch_xml.c:1329 Couldnt open
>>>     /opt/freeswitch/conf/lang/fr/*.xml (No such file or directory)
>>>     Error including /opt/freeswitch/conf/lang/ru/*.xml (No such file
>>>     or directory)
>>>     2011-06-15 09:00:18.008426 [ERR] switch_xml.c:1329 Couldnt open
>>>     /opt/freeswitch/conf/lang/ru/*.xml (No such file or directory)
>>>     2011-06-15 09:00:18.021491 [INFO] mod_enum.c:808 ENUM Reloaded
>>>     2011-06-15 09:00:18.021491 [INFO] switch_time.c:950 Timezone
>>>     reloaded 530 definitions
>>>     2011-06-15 09:00:18.022839 [CRIT] switch_loadable_module.c:926
>>>     Error Loading module /opt/freeswitch/mod/mod_lua.so
>>>     **/opt/freeswitch/mod/mod_lua.so: cannot open shared object
>>>     file: No such file or directory**
>>>
>>>     -ERR unloading module [No such module!]
>>>     +OK Reloading XML
>>>     -ERR loading module [module load file routine returned an error]
>>>
>>>     On 6/14/2011 4:55 PM, Avi Marcus wrote:
>>>>     What happens when you run "reload mod_lua" from the CLI?
>>>>
>>>>     -Avi Marcus
>>>>
>>>>
>>>>     On Wed, Jun 15, 2011 at 12:48 AM, Wes <wes-fs at 499x.com
>>>>     <mailto:wes-fs at 499x.com>> wrote:
>>>>
>>>>         I'm getting "invalid application lua" however, in
>>>>         modules.conf, the line
>>>>         for <load module="mod_lua"/> is active and not commented out.
>>>>
>>>>         Any help is appreciated. Thanks.
>>>>
>>>>         _______________________________________________
>>>>         FreeSWITCH-users mailing list
>>>>         FreeSWITCH-users at lists.freeswitch.org
>>>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>         http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>
>>>>
>>>>
>>>>     _______________________________________________
>>>>     FreeSWITCH-users mailing list
>>>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org  <http://www.freeswitch.org/>
>>>
>>>     _______________________________________________
>>>     FreeSWITCH-users mailing list
>>>     FreeSWITCH-users at lists.freeswitch.org
>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From daldworth at teliax.com  Wed Jun 15 08:24:23 2011
From: daldworth at teliax.com (David Aldworth)
Date: Tue, 14 Jun 2011 22:24:23 -0600
Subject: [Freeswitch-users] 183 after 180
Message-ID: <9AD5D885-6663-4C1D-B3FA-7CFC1C49F68F@teliax.com>

I am trying to find the best solution to generate ring on 180 but bypass on 183. We are in the middle, customer -> FS -> pstn. We bypass_media=true for all calls. However, sometimes we get a 180 from the PSTN and FS relays that to the customer, but the customer is not generating ring so the calling party hears dead air. So, the solution would be to anchor media and generate ring when we get a 180, but once the 183 or 200 hits, bypass starts. I suppose we could do an expression match on the 180, ignore early media, generate ring, and bypass media after bridge. Just wondering if there was a better alternative solution. 

From qingquan at globalroam.com  Wed Jun 15 11:23:37 2011
From: qingquan at globalroam.com (qingquan luo)
Date: Wed, 15 Jun 2011 15:23:37 +0800
Subject: [Freeswitch-users] in bypass_media mode,
 2833 payload type modified by freeswitch!
In-Reply-To: <4A5154F6-90DE-4B40-9F24-E3E0D0824D02@ipeva.fr>
References: <BANLkTi=HpPE7q4Kbrkwe-KJxj69SYJtx0Q@mail.gmail.com>
	<4A5154F6-90DE-4B40-9F24-E3E0D0824D02@ipeva.fr>
Message-ID: <BANLkTikfoYbL-atXM78W7BTLe3LNUheF9A@mail.gmail.com>

Hi David,
     Thank you for reminder,   I resend this mail in mail list, Because I
can not reply preview mail( I make a mistake to subscribe this mail list in
Daily Digest mode.)
     So I need resend this mail for require help.

For this issue, I has paste sip debug output in:

information with "sofia global siptrace on"
http://pastebin.freeswitch.org/16496


 Thanks



On Wed, Jun 15, 2011 at 3:15 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> You have a mail issue. Check your spam...
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 15/06/2011 ? 03:55, qingquan luo a ?crit :
>
> Hi All,
>
>   I use freeswitch to bridge one incoming call to other target phone
> number.
>   I use bypass_media mode.
>   So the rtp is not go through the freeswitch,
>   By I notice that When LegB reply 183 or 200 SDP message. Freeswitch
> change it 2833 telephone-event payload type. and forwarding the message to
> caller. This make the caller use wrong dtmf payload type to send dtmf
>
>   What wrong with that?  How can fix it?
>   Any information or help is welcome.
>
> information with "sofia global siptrace on"
> http://pastebin.freeswitch.org/16496
>
> Thanks
>
>
> Best Regards
>
>
> Qingquan
>
> --
> Using Gmail? Please read this important notice:
> http://www.fsf.org/campaigns/jstrap/gmail?40922.
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Using Gmail? Please read this important notice:
http://www.fsf.org/campaigns/jstrap/gmail?40922.
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From doreme202002 at yahoo.com  Wed Jun 15 13:36:39 2011
From: doreme202002 at yahoo.com (hala alramli)
Date: Wed, 15 Jun 2011 10:36:39 +0100 (BST)
Subject: [Freeswitch-users] Fw: caller phone hangup call after callee answer
Message-ID: <34394.27154.qm@web24604.mail.ird.yahoo.com>

i have kamailio server with freeswitch ?but i face this problem .any help to explain the reason.
i have a problem here ?that when callee answer caller phone send bye and terminate the call i will attach the track.
?
freeswitch listen to?sip_server:5068


U 2011/06/13 13:23:41.764946 caller_ip_XXXXX:1024 -> sip_server:5060
INVITE sip:callee at sip..XXXX.XXXSIP/2.0.
Via: SIP/2.0/UDP
192.168.1.130:5060;branch=z9hG4bK-a9c2d8ca.
From: <sip:909999999 at sip.XXXX.XXX>;tag=4fde77c876fbda1ao0.
To: <sip:callee at sip.XXXX.XXX>.
Call-ID: 850573e0-95b74ce2 at 192.168.1.130.
CSeq: 101 INVITE.
Max-Forwards: 70.
Contact: <sip:909999999 at 192.168.1.130:5060>.
Expires: 240.
User-Agent: Linksys/PAP2T-3.1.15(LS).
Content-Length: 444.
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS,
REFER.
Supported: x-sipura.
Content-Type: application/sdp.
.
v=0.
o=- 785603 785603 IN IP4 192.168.1.130.
s=-.
c=IN IP4 192.168.1.130.
t=0 0.
m=audio 16416 RTP/AVP 18 0 2 4 8 96 97 98 100 101.
a=rtpmap:18 G729a/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:2 G726-32/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:96 G726-40/8000.
a=rtpmap:97 G726-24/8000.
a=rtpmap:98 G726-16/8000.
a=rtpmap:100 NSE/8000.
a=fmtp:100 192-193.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:30.
a=sendrecv.
?
?
U 2011/06/13 13:23:41.765588 sip_server:5060 -> caller_ip_XXXXX:1024
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP
192.168.1.130:5060;branch=z9hG4bK-a9c2d8ca;rport=1024;received=caller_ip_XXXXX.
From: <sip:909999999 at sip.XXXX.XXX>;tag=4fde77c876fbda1ao0.
To: <sip:callee at sip.XXXX.XXX>;tag=4471d251144740103ef2d5e5631bdc66.e595.
Call-ID: 850573e0-95b74ce2 at 192.168.1.130.
CSeq: 101 INVITE.
Proxy-Authenticate: Digest realm="sip.XXXX.XXX",
nonce="4df5f4690000f03a53f01a12380d236e373437474bf441ef".
Server: Kamailio (1.4.4-notls (x86_64/linux)).
Content-Length: 0.
.
?
?
U 2011/06/13 13:23:41.822266 caller_ip_XXXXX:1024 -> sip_server:5060
ACK sip:callee at sip..XXXX.XXXSIP/2.0.
Via: SIP/2.0/UDP
192.168.1.130:5060;branch=z9hG4bK-a9c2d8ca.
From: <sip:909999999 at sip.XXXX.XXX>;tag=4fde77c876fbda1ao0.
To: <sip:callee at sip.XXXX.XXX>;tag=4471d251144740103ef2d5e5631bdc66.e595.
Call-ID: 850573e0-95b74ce2 at 192.168.1.130.
CSeq: 101 ACK.
Max-Forwards: 70.
Contact: <sip:909999999 at 192.168.1.130:5060>.
User-Agent: Linksys/PAP2T-3.1.15(LS).
Content-Length: 0.
.
?
?
U 2011/06/13 13:23:41.834779 caller_ip_XXXXX:1024 -> sip_server:5060
INVITE sip:callee at sip..XXXX.XXXSIP/2.0.
Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK-9f4eddee.
From: <sip:909999999 at sip.XXXX.XXX>;tag=4fde77c876fbda1ao0.
To: <sip:callee at sip.XXXX.XXX>.
Call-ID: 850573e0-95b74ce2 at 192.168.1.130.
CSeq: 102 INVITE.
Max-Forwards: 70.
Proxy-Authorization: Digest username="909999999",realm="sip.XXXX.XXX",nonce="4df5f4690000f03a53f01a12380d236e373437474bf441
ef",uri="sip:callee at sip.XXXX.XXX",algorithm=MD5,response="82a5c2648544cf1abd616f7816a61a5f".
Contact: <sip:909999999 at 192.168.1.130:5060>.
Expires: 240.
User-Agent: Linksys/PAP2T-3.1.15(LS).
Content-Length: 444.
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS,
REFER.
Supported: x-sipura.
Content-Type: application/sdp.
.
v=0.
o=- 785603 785603 IN IP4 192.168.1.130.
s=-.
c=IN IP4 192.168.1.130.
t=0 0.
m=audio 16416 RTP/AVP 18 0 2 4 8 96 97 98 100 101.
a=rtpmap:18 G729a/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:2 G726-32/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:96 G726-40/8000.
a=rtpmap:97 G726-24/8000.
a=rtpmap:98 G726-16/8000.
a=rtpmap:100 NSE/8000.
a=fmtp:100 192-193.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:30.
a=sendrecv.
?
?
U 2011/06/13 13:23:41.838496 sip_server:5060 -> caller_ip_XXXXX:1024
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP
192.168.1.130:5060;branch=z9hG4bK-9f4eddee;rport=1024;received=caller_ip_XXXXX.
From: <sip:909999999 at sip.XXXX.XXX>;tag=4fde77c876fbda1ao0.
To: <sip:callee at sip.XXXX.XXX>.
Call-ID: 850573e0-95b74ce2 at 192.168.1.130.
CSeq: 102 INVITE.
Server: Kamailio (1.4.4-notls (x86_64/linux)).
Content-Length: 0.
.
?
?
U 2011/06/13 13:23:41.843647 sip_server:5068 -> gateway_XXXX:5060
INVITE sip:122callee at gateway_XXXX
SIP/2.0.
Via: SIP/2.0/UDP sip_server:5068;rport;branch=z9hG4bKcZmy3ypB0jm1B.
Max-Forwards: 14.
From: "909999999" <sip:909999999 at sip.XXXX.XXX>;tag=5SNSNaj6XQ8Kr.
To: <sip:122callee at gateway_XXXX>.
Call-ID: 6f179185-1052-122f-49bf-00221988c79d.
CSeq: 13646046 INVITE.
Contact: <sip:ufs at sip_server:5068>.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11078.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,
SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer.
Min-SE: 120.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 478.
P-PInfo: 0.
Remote-Party-ID: "909999999" <sip:909999999 at sip.XXXX.XXX>;screen=yes;privacy=off.
.
v=0.
o=-
4131904911551689921 6268338771912655224 IN IP4 sip_server.
s=-.
c=IN IP4 sip_server.
t=0 0.
m=audio 19594 RTP/AVP 18 0 2 4 8 96 97 98 100 101.
a=rtpmap:18 G729a/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:2 G726-32/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:96 G726-40/8000.
a=rtpmap:97 G726-24/8000.
a=rtpmap:98 G726-16/8000.
a=rtpmap:100 NSE/8000.
a=fmtp:100 192-193.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:30.
a=direction:active.
?
?
U 2011/06/13 13:23:41.847833 gateway_XXXX:5060 -> sip_server:5068
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP sip_server:5068;received=sip_server;rport=5068;branch=z9hG4bKcZmy3ypB0jm1B.
From: "909999999" <sip:909999999 at sip.XXXX.XXX>;tag=5SNSNaj6XQ8Kr.
To: <sip:122callee at gateway_XXXX>.
Call-ID: 6f179185-1052-122f-49bf-00221988c79d.
CSeq: 13646046 INVITE.
Content-Length: 0.
.
?
?
U 2011/06/13 13:23:47.038040 gateway_XXXX:5060 -> sip_server:5068
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP sip_server:5068;received=sip_server;rport=5068;branch=z9hG4bKcZmy3ypB0jm1B.
From: "909999999" <sip:909999999 at sip.XXXX.XXX>;tag=5SNSNaj6XQ8Kr.
To: <sip:122callee at gateway_XXXX>;tag=SDlfkl999-e4e6842982432011613141134.
Call-ID: 6f179185-1052-122f-49bf-00221988c79d.
CSeq: 13646046 INVITE.
Server: CS2000_NGSS/8.0.
Supported: 100rel.
Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK.
Contact: <sip:gateway_XXXX:5060;transport=UDP>.
Content-Type: application/sdp.
Content-Length: 246.
.
v=0.
o=PVG 1307966093710 1307966093710 IN IP4 gateway_XXXX.
s=-.
p=+1 6135555555.
c=IN IP4 gateway_XXXX.
t=0 0.
m=audio 36568 RTP/AVP 18 101.
a=fmtp:18 annexb=no.
a=fmtp:101 0-15.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:30.
?
?
U 2011/06/13 13:23:47.049924 sip_server:5060 -> caller_ip_XXXXX:1024
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP 192.168.1.130:5060;rport=1024;received=caller_ip_XXXXX;branch=z9hG4bK-9f4eddee.
Record-Route: <sip:sip_server;lr=on;ftag=4fde77c876fbda1ao0>.
From: <sip:909999999 at sip.XXXX.XXX>;tag=4fde77c876fbda1ao0.
To: <sip:callee at sip.XXXX.XXX>;tag=4gv0KF120ej1c.
Call-ID: 850573e0-95b74ce2 at 192.168.1.130.
CSeq: 102 INVITE.
Contact: <sip:mod_sofia at sip_server:5068;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11078.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,
SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 248.
.
v=0.
o=PVG
2756959439342557186 2159116678429256306 IN IP4 sip_server.
s=-.
p=+1 6135555555.
c=IN IP4 sip_server.
t=0 0.
m=audio 0 RTP/AVP 99 101.
a=rtpmap:99 G729/8000.
a=fmtp:99 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
?
?
U 2011/06/13 13:23:49.770967 gateway_XXXX:5060 -> sip_server:5068
SIP/2.0 200 OK.
Via: SIP/2.0/UDP sip_server:5068;received=sip_server;rport=5068;branch=z9hG4bKcZmy3ypB0jm1B.
From: "909999999" <sip:909999999 at sip.XXXX.XXX>;tag=5SNSNaj6XQ8Kr.
To: <sip:122callee at gateway_XXXX>;tag=SDlfkl999-e4e6842982432011613141134.
Call-ID: 6f179185-1052-122f-49bf-00221988c79d.
CSeq: 13646046 INVITE.
Server: CS2000_NGSS/8.0.
Supported: 100rel.
Allow:
ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK.
Contact: <sip:gateway_XXXX:5060;transport=UDP>.
Content-Type: application/sdp.
Content-Length: 246.
.
v=0.
o=PVG 1307966093710 1307966093710 IN IP4 gateway_XXXX.
s=-.
p=+1 6135555555.
c=IN IP4 gateway_XXXX.
t=0 0.
m=audio 36568 RTP/AVP 18 101.
a=fmtp:18 annexb=no.
a=fmtp:101 0-15.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:30.
?
?
U 2011/06/13 13:23:49.771167 sip_server:5068 -> gateway_XXXX:5060
ACK sip:gateway_XXXX:5060;transport=UDP SIP/2.0.
Via: SIP/2.0/UDP sip_server:5068;rport;branch=z9hG4bKD8DQ5S7eXUamQ.
Max-Forwards: 70.
From: "909999999" <sip:909999999 at sip.XXXX.XXX>;tag=5SNSNaj6XQ8Kr.
To: <sip:122callee at gateway_XXXX>;tag=SDlfkl999-e4e6842982432011613141134.
Call-ID: 6f179185-1052-122f-49bf-00221988c79d.
CSeq: 13646046 ACK.
Contact: <sip:ufs at sip_server:5068>.
Content-Length: 0.
.
?
?
U 2011/06/13 13:23:49.774216 sip_server:5060 -> caller_ip_XXXXX:1024
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.1.130:5060;rport=1024;received=caller_ip_XXXXX;branch=z9hG4bK-9f4eddee.
Record-Route: <sip:sip_server;lr=on;ftag=4fde77c876fbda1ao0>.
From: <sip:909999999 at sip.XXXX.XXX>;tag=4fde77c876fbda1ao0.
To: <sip:callee at sip.XXXX.XXX>;tag=4gv0KF120ej1c.
Call-ID: 850573e0-95b74ce2 at 192.168.1.130.
CSeq: 102 INVITE.
Contact: <sip:mod_sofia at sip_server:5068;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11078.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,
SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer.
Min-SE: 120.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 248.
.
v=0.
o=PVG
2756959439342557186 2159116678429256306 IN IP4 sip_server.
s=-.
p=+1 6135555555.
c=IN IP4 sip_server.
t=0 0.
m=audio 0 RTP/AVP 99 101.
a=rtpmap:99 G729/8000.
a=fmtp:99 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
?
?
U 2011/06/13 13:23:49.836231 caller_ip_XXXXX:1024 -> sip_server:5060
ACK sip:mod_sofia at sip_server:5068;transport=udp SIP/2.0.
Via: SIP/2.0/UDP
192.168.1.130:5060;branch=z9hG4bK-d84f3d54.
From: <sip:909999999 at sip.XXXX.XXX>;tag=4fde77c876fbda1ao0.
To: <sip:callee at sip.XXXX.XXX>;tag=4gv0KF120ej1c.
Call-ID: 850573e0-95b74ce2 at 192.168.1.130.
CSeq: 102 ACK.
Max-Forwards: 70.
Route: <sip:sip_server;lr=on;ftag=4fde77c876fbda1ao0>.
Proxy-Authorization: Digest username="909999999",realm="sip.XXXX.XXX",nonce="4df5f4690000f03a53f01a12380d236e373437474bf441
ef",uri="sip:mod_sofia at sip_server:5068",algorithm=MD5,response="1dc1c1666a5d210779ab1daf84b6eb10".
Contact: <sip:909999999 at 192.168.1.130:5060>.
User-Agent: Linksys/PAP2T-3.1.15(LS).
Content-Length: 0.
.
?
?
U 2011/06/13 13:23:49.843388 caller_ip_XXXXX:1024 -> sip_server:5060
BYE sip:mod_sofia at sip_server:5068;transport=udp SIP/2.0.
Via: SIP/2.0/UDP
192.168.1.130:5060;branch=z9hG4bK-1710f527.
From: <sip:909999999 at sip.XXXX.XXX>;tag=4fde77c876fbda1ao0.
To: <sip:callee at sip.XXXX.XXX>;tag=4gv0KF120ej1c.
Call-ID: 850573e0-95b74ce2 at 192.168.1.130.
CSeq: 103 BYE.
Max-Forwards: 70.
Route: <sip:sip_server;lr=on;ftag=4fde77c876fbda1ao0>.
Proxy-Authorization: Digest username="909999999",realm="sip.XXXX.XXX",nonce="4df5f4690000f03a53f01a12380d236e373437474bf441
ef",uri="sip:mod_sofia at sip_server:5068",algorithm=MD5,response="ff113ffd721bbaef656458c39840e450".
User-Agent: Linksys/PAP2T-3.1.15(LS).
Content-Length: 0.
.
?
?
U 2011/06/13 13:23:49.845476 sip_server:5060 -> caller_ip_XXXXX:1024
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.1.130:5060;rport=1024;received=caller_ip_XXXXX;branch=z9hG4bK-1710f527.
From: <sip:909999999 at sip.XXXX.XXX>;tag=4fde77c876fbda1ao0.
To: <sip:callee at sip.XXXX.XXX>;tag=4gv0KF120ej1c.
Call-ID: 850573e0-95b74ce2 at 192.168.1.130.
CSeq: 103 BYE.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11078.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,
SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
Supported: timer, precondition, path, replaces.
Content-Length: 0.
.
?
?
U 2011/06/13 13:23:49.848748 sip_server:5068 -> gateway_XXXX:5060
BYE sip:gateway_XXXX:5060;transport=UDP SIP/2.0.
Via: SIP/2.0/UDP sip_server:5068;rport;branch=z9hG4bKeH7F7mrjt406j.
Max-Forwards: 70.
From: "909999999" <sip:909999999 at sip.XXXX.XXX>;tag=5SNSNaj6XQ8Kr.
To: <sip:122callee at gateway_XXXX>;tag=SDlfkl999-e4e6842982432011613141134.
Call-ID: 6f179185-1052-122f-49bf-00221988c79d.
CSeq: 13646047 BYE.
Contact: <sip:ufs at sip_server:5068>.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11078.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,
SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
Supported: timer, precondition, path, replaces.
Reason: Q.850;cause=16;text="NORMAL_CLEARING".
Content-Length: 0.
.
?
?
U 2011/06/13 13:23:49.858691 gateway_XXXX:5060 -> sip_server:5068
SIP/2.0 200 OK.
Via: SIP/2.0/UDP sip_server:5068;received=sip_server;rport=5068;branch=z9hG4bKeH7F7mrjt406j.
From: "909999999" <sip:909999999 at sip.XXXX.XXX>;tag=5SNSNaj6XQ8Kr.
To: <sip:122callee at gateway_XXXX>;tag=SDlfkl999-e4e6842982432011613141134.
Call-ID: 6f179185-1052-122f-49bf-00221988c79d.
CSeq: 13646047 BYE.
Server: CS2000_NGSS/8.0.
Supported: 100rel.
Allow:
ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK.
Content-Length: 0.
.
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From Stefan.Weigel at allianz-warranty.com  Wed Jun 15 19:14:08 2011
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Wed, 15 Jun 2011 17:14:08 +0200
Subject: [Freeswitch-users] unixODBC MySQL mysqld-5.0.77 - error messages
	with mod_callcenter
Message-ID: <5003D7D3E06F514E8C682F18D223265C04D3B36C29@AZWSMS03.azwarranty.int>

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From nsirugudi at gmail.com  Wed Jun 15 07:20:40 2011
From: nsirugudi at gmail.com (Narendra Sirugudi)
Date: Wed, 15 Jun 2011 08:50:40 +0530
Subject: [Freeswitch-users] Originating a call from a confeence is not
 working ?
In-Reply-To: <BANLkTinC6WadzA41QH7fvXu-w41cv5oh7Q@mail.gmail.com>
References: <BANLkTi=_ogA9pig3HTQCoG2mWB9P_a_8ng@mail.gmail.com>
	<BANLkTinC6WadzA41QH7fvXu-w41cv5oh7Q@mail.gmail.com>
Message-ID: <BANLkTimhSGqaay=hK9rrb8g=xPeD3AhoBQ@mail.gmail.com>

Hi,
Thanks for your reply.

But your suggestion did not work.

What worked is:
freeswitch at internal> conference 3001-135.254.210.166 dial
{originate_timeout=30}sofia/internal/1005%135.254.210.166 1234567890
FreeSWITCH_Conference

The change is instead of 1005 at 135.254.210.166, i now have
1005%135.254.210.166
i.e. "@" replaced with "%".

this works fine.

thanks again,
--naren



On Mon, Jun 13, 2011 at 8:57 PM, Michael Collins <msc at freeswitch.org> wrote:

> Just curious - Try this and let us know if it behaves differently:
>
> conference 3001-10.10.10.16 dial {originate_timeout=30}<%7Boriginate_timeout=30%7Dsofia/internal/1005 at 135.254.210.166>user/1005 1234567890
> FreeSWITCH_Conference
>
> -MC
>
>   On Sun, Jun 12, 2011 at 8:32 AM, Narendra Sirugudi <nsirugudi at gmail.com>wrote:
>
>>   Hi,
>>
>> I am trying to use the dial-out feature of the conference application of
>> freeswitch.
>> I am using freeswtich version 1.0.6.
>>
>> As i understand the conference gets created when the first user joins.
>> Hence i made one user with number 1004 join the conference.
>>
>> freeswitch at internal> conference list
>> Conference 3001-10.10.10.16 (1 member rate: 8000)
>> 6;sofia/internal/1004 at 10.10.10.16
>> ;08178796-a06e-4ee8-81fb-1f0cc53f8fb5;1004;1004;hear|speak|floor;0;0;300
>>
>> Now i dail out another user 1005 using the command :
>> freeswitch at internal> conference 3001-10.10.10.16 dial
>> {originate_timeout=30}sofia/internal/1005 at 10.10.10.16<%7Boriginate_timeout=30%7Dsofia/internal/1005 at 135.254.210.166>1234567890 FreeSWITCH_Conference
>> This does not work. I observe the following errors in the fs_cli logs:
>>
>> *2011-06-12 10:57:57.621014 [ERR] sofia.c:5366 Cannot Blind Transfer 1
>> Legged calls*
>> can anyone tell what could be going wrong ?
>>
>> thanks
>> --naren
>>
>> The complete logs are given below:
>>
>> #############################################################
>> freeswitch at internal> conference 3001-10.10.10.16 dial
>> {originate_timeout=30}sofia/internal/1005 at 10.10.10.16 1234567890
>> FreeSWITCH_Conference
>> Call Requested: result: [NO_USER_RESPONSE]
>> 2011-06-12 10:57:57.606002 [DEBUG] switch_ivr_originate.c:1885 variable
>> string 0 = [ignore_early_media=true]
>> freeswitch at internal> 2011-06-12 10:57:57.606002 [DEBUG]
>> switch_ivr_originate.c:1885 variable string 1 = [originate_timeout=30]
>> 2011-06-12 10:57:57.606002 [NOTICE] switch_channel.c:669 New Channel
>> sofia/internal/1005 at 10.10.10.16 [a77137ad-ad1a-45e9-ac27-88a996cd865d]
>> 2011-06-12 10:57:57.606002 [DEBUG] mod_sofia.c:3384 (
>> sofia/internal/1005 at 10.10.10.16) State Change CS_NEW -> CS_INIT
>> 2011-06-12 10:57:57.606002 [DEBUG] switch_core_session.c:1021 Send signal
>> sofia/internal/1005 at 10.10.10.16 [BREAK]
>> 2011-06-12 10:57:57.607036 [DEBUG] switch_core_state_machine.c:314 (
>> sofia/internal/1005 at 10.10.10.16) Running State Change CS_INIT
>> 2011-06-12 10:57:57.607036 [DEBUG] switch_core_state_machine.c:338 (
>> sofia/internal/1005 at 10.10.10.16) State INIT
>> 2011-06-12 10:57:57.607036 [DEBUG] mod_sofia.c:83
>> sofia/internal/1005 at 10.10.10.16 SOFIA INIT
>> 2011-06-12 10:57:57.607036 [DEBUG] mod_sofia.c:117 (
>> sofia/internal/1005 at 10.10.10.16) State Change CS_INIT -> CS_ROUTING
>> 2011-06-12 10:57:57.607036 [DEBUG] switch_core_session.c:1021 Send signal
>> sofia/internal/1005 at 10.10.10.16 [BREAK]
>> 2011-06-12 10:57:57.607036 [DEBUG] switch_core_state_machine.c:338 (
>> sofia/internal/1005 at 10.10.10.16) State INIT going to sleep
>> 2011-06-12 10:57:57.607036 [DEBUG] switch_core_state_machine.c:314 (
>> sofia/internal/1005 at 10.10.10.16) Running State Change CS_ROUTING
>> 2011-06-12 10:57:57.607036 [DEBUG] switch_core_state_machine.c:341 (
>> sofia/internal/1005 at 10.10.10.16) State ROUTING
>> 2011-06-12 10:57:57.607036 [DEBUG] mod_sofia.c:140
>> sofia/internal/1005 at 10.10.10.16 SOFIA ROUTING
>> 2011-06-12 10:57:57.607036 [DEBUG] switch_ivr_originate.c:66 (
>> sofia/internal/1005 at 10.10.10.16) State Change CS_ROUTING ->
>> CS_CONSUME_MEDIA
>> 2011-06-12 10:57:57.607036 [DEBUG] switch_core_session.c:1021 Send signal
>> sofia/internal/1005 at 10.10.10.16 [BREAK]
>> 2011-06-12 10:57:57.607036 [DEBUG] switch_core_state_machine.c:341 (
>> sofia/internal/1005 at 10.10.10.16) State ROUTING going to sleep
>> 2011-06-12 10:57:57.607036 [DEBUG] switch_core_state_machine.c:314 (
>> sofia/internal/1005 at 10.10.10.16) Running State Change CS_CONSUME_MEDIA
>> 2011-06-12 10:57:57.607036 [DEBUG] switch_core_state_machine.c:360 (
>> sofia/internal/1005 at 10.10.10.16) State CONSUME_MEDIA
>> 2011-06-12 10:57:57.607036 [DEBUG] switch_core_state_machine.c:360 (
>> sofia/internal/1005 at 10.10.10.16) State CONSUME_MEDIA going to sleep
>> 2011-06-12 10:57:57.608051 [DEBUG] sofia.c:4153 Channel
>> sofia/internal/1005 at 10.10.10.16 entering state [calling][0]
>> 2011-06-12 10:57:57.608051 [DEBUG] sofia.c:5847 IP 10.10.10.16 Rejected by
>> acl "domains". Falling back to Digest auth.
>> 2011-06-12 10:57:57.608051 [NOTICE] switch_channel.c:669 New Channel
>> sofia/internal/1234567890 at 10.10.10.16[86e5b4a9-8c63-4d22-b087-94c4b95d1abb]
>> 2011-06-12 10:57:57.609139 [DEBUG] switch_core_state_machine.c:314 (
>> sofia/internal/1234567890 at 10.10.10.16) Running State Change CS_NEW
>> 2011-06-12 10:57:57.609139 [DEBUG] switch_core_state_machine.c:320 (
>> sofia/internal/1234567890 at 10.10.10.16) State NEW
>> 2011-06-12 10:57:57.616761 [DEBUG] sofia.c:4153 Channel
>> sofia/internal/1234567890 at 10.10.10.16 entering state [received][100]
>> 2011-06-12 10:57:57.616761 [DEBUG] sofia.c:4164 Remote SDP:
>> v=0
>> o=FreeSWITCH 1307870225 1307870226 IN IP4 10.10.10.16
>> s=FreeSWITCH
>> c=IN IP4 10.10.10.16
>> t=0 0
>> m=audio 20452 RTP/AVP 115 107 9 0 8 3 101 13
>> a=rtpmap:115 G7221/32000
>> a=fmtp:115 bitrate=48000
>> a=rtpmap:107 G7221/16000
>> a=fmtp:107 bitrate=32000
>> a=rtpmap:9 G722/8000
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:3 GSM/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=rtpmap:13 CN/8000
>> a=ptime:20
>> 2011-06-12 10:57:57.616761 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
>> [G7221:115:32000:20]/[G7221:115:32000:20]
>> 2011-06-12 10:57:57.616761 [DEBUG] sofia_glue.c:2354 Set Codec
>> sofia/internal/1234567890 at 10.10.10.16 G7221/32000 20 ms 640 samples
>> 2011-06-12 10:57:57.617820 [DEBUG] sofia_glue.c:3524 Set 2833 dtmf
>> send/recv payload to 101
>> 2011-06-12 10:57:57.617820 [DEBUG] sofia.c:4310 (
>> sofia/internal/1234567890 at 10.10.10.16) State Change CS_NEW -> CS_INIT
>> 2011-06-12 10:57:57.617820 [DEBUG] switch_core_session.c:1021 Send signal
>> sofia/internal/1234567890 at 10.10.10.16 [BREAK]
>> 2011-06-12 10:57:57.618925 [DEBUG] switch_core_state_machine.c:314 (
>> sofia/internal/1234567890 at 10.10.10.16) Running State Change CS_INIT
>> 2011-06-12 10:57:57.618925 [DEBUG] switch_core_state_machine.c:338 (
>> sofia/internal/1234567890 at 10.10.10.16) State INIT
>> 2011-06-12 10:57:57.618925 [DEBUG] mod_sofia.c:83
>> sofia/internal/1234567890 at 10.10.10.16 SOFIA INIT
>> 2011-06-12 10:57:57.618925 [DEBUG] mod_sofia.c:117 (
>> sofia/internal/1234567890 at 10.10.10.16) State Change CS_INIT -> CS_ROUTING
>> 2011-06-12 10:57:57.618925 [DEBUG] switch_core_session.c:1021 Send signal
>> sofia/internal/1234567890 at 10.10.10.16 [BREAK]
>> 2011-06-12 10:57:57.618925 [DEBUG] switch_core_state_machine.c:338 (
>> sofia/internal/1234567890 at 10.10.10.16) State INIT going to sleep
>> 2011-06-12 10:57:57.618925 [DEBUG] switch_core_state_machine.c:314 (
>> sofia/internal/1234567890 at 10.10.10.16) Running State Change CS_ROUTING
>> 2011-06-12 10:57:57.618925 [DEBUG] switch_core_state_machine.c:341 (
>> sofia/internal/1234567890 at 10.10.10.16) State ROUTING
>> 2011-06-12 10:57:57.618925 [DEBUG] mod_sofia.c:140
>> sofia/internal/1234567890 at 10.10.10.16 SOFIA ROUTING
>> 2011-06-12 10:57:57.618925 [DEBUG] switch_core_state_machine.c:77
>> sofia/internal/1234567890 at 10.10.10.16 Standard ROUTING
>> 2011-06-12 10:57:57.618925 [INFO] mod_dialplan_xml.c:418 Processing
>> FreeSWITCH_Conference->1005 in context public
>> Dialplan: sofia/internal/1234567890 at 10.10.10.16 parsing [public->unloop]
>> continue=false
>> Dialplan: sofia/internal/1234567890 at 10.10.10.16 Regex (PASS) [unloop]
>> ${unroll_loops}(true) =~ /^true$/ break=on-false
>> Dialplan: sofia/internal/1234567890 at 10.10.10.16 Regex (PASS) [unloop]
>> ${sip_looped_call}(true) =~ /^true$/ break=on-false
>> Dialplan: sofia/internal/1234567890 at 10.10.10.16 Action
>> deflect(${destination_number})
>> 2011-06-12 10:57:57.620002 [DEBUG] switch_core_state_machine.c:119 (
>> sofia/internal/1234567890 at 10.10.10.16) State Change CS_ROUTING ->
>> CS_EXECUTE
>> 2011-06-12 10:57:57.620002 [DEBUG] switch_core_session.c:1021 Send signal
>> sofia/internal/1234567890 at 10.10.10.16 [BREAK]
>> 2011-06-12 10:57:57.620002 [DEBUG] switch_core_state_machine.c:341 (
>> sofia/internal/1234567890 at 10.10.10.16) State ROUTING going to sleep
>> 2011-06-12 10:57:57.620002 [DEBUG] switch_core_state_machine.c:314 (
>> sofia/internal/1234567890 at 10.10.10.16) Running State Change CS_EXECUTE
>> 2011-06-12 10:57:57.620002 [DEBUG] switch_core_state_machine.c:348 (
>> sofia/internal/1234567890 at 10.10.10.16) State EXECUTE
>> 2011-06-12 10:57:57.620002 [DEBUG] mod_sofia.c:226
>> sofia/internal/1234567890 at 10.10.10.16 SOFIA EXECUTE
>> 2011-06-12 10:57:57.620002 [DEBUG] switch_core_state_machine.c:157
>> sofia/internal/1234567890 at 10.10.10.16 Standard EXECUTE
>> EXECUTE sofia/internal/1234567890 at 10.10.10.16 deflect(1005)
>> 2011-06-12 10:57:57.621014 [DEBUG] sofia.c:5004 Process REFER to [
>> 1005 at 10.10.10.16]
>> *2011-06-12 10:57:57.621014 [ERR] sofia.c:5366 Cannot Blind Transfer 1
>> Legged calls*
>> 2011-06-12 10:57:57.622041 [DEBUG] sofia.c:4153 Channel
>> sofia/internal/1005 at 10.10.10.16 entering state [terminated][480]
>> 2011-06-12 10:57:57.622041 [NOTICE] sofia.c:4789 Hangup
>> sofia/internal/1005 at 10.10.10.16 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE]
>> 2011-06-12 10:57:57.622041 [DEBUG] switch_channel.c:2102 Send signal
>> sofia/internal/1005 at 10.10.10.16 [KILL]
>> 2011-06-12 10:57:57.622041 [DEBUG] switch_core_session.c:1021 Send signal
>> sofia/internal/1005 at 10.10.10.16 [BREAK]
>> 2011-06-12 10:57:57.622041 [DEBUG] switch_core_state_machine.c:314 (
>> sofia/internal/1005 at 10.10.10.16) Running State Change CS_HANGUP
>> 2011-06-12 10:57:57.623126 [DEBUG] switch_core_state_machine.c:499 (
>> sofia/internal/1005 at 10.10.10.16) State HANGUP
>> 2011-06-12 10:57:57.623126 [DEBUG] mod_sofia.c:408
>> sofia/internal/1005 at 10.10.10.16 Overriding SIP cause 408 with 480 from
>> the other leg
>> 2011-06-12 10:57:57.623126 [DEBUG] mod_sofia.c:414 Channel
>> sofia/internal/1005 at 10.10.10.16 hanging up, cause: NO_USER_RESPONSE
>> 2011-06-12 10:57:57.623126 [DEBUG] switch_ivr_originate.c:3228 Originate
>> Resulted in Error Cause: 18 [NO_USER_RESPONSE]
>> 2011-06-12 10:57:57.636515 [DEBUG] switch_core_state_machine.c:46
>> sofia/internal/1005 at 10.10.10.16 Standard HANGUP, cause: NO_USER_RESPONSE
>> 2011-06-12 10:57:57.636515 [DEBUG] switch_core_state_machine.c:499 (
>> sofia/internal/1005 at 10.10.10.16) State HANGUP going to sleep
>> 2011-06-12 10:57:57.636515 [DEBUG] switch_core_state_machine.c:333 (
>> sofia/internal/1005 at 10.10.10.16) State Change CS_HANGUP -> CS_REPORTING
>> 2011-06-12 10:57:57.636515 [DEBUG] switch_core_session.c:1021 Send signal
>> sofia/internal/1005 at 10.10.10.16 [BREAK]
>> 2011-06-12 10:57:57.637437 [DEBUG] switch_core_state_machine.c:314 (
>> sofia/internal/1005 at 10.10.10.16) Running State Change CS_REPORTING
>> 2011-06-12 10:57:57.637437 [DEBUG] switch_core_state_machine.c:590 (
>> sofia/internal/1005 at 10.10.10.16) State REPORTING
>> 2011-06-12 10:57:57.637437 [DEBUG] switch_core_state_machine.c:53
>> sofia/internal/1005 at 10.10.10.16 Standard REPORTING, cause:
>> NO_USER_RESPONSE
>> 2011-06-12 10:57:57.637437 [DEBUG] switch_core_state_machine.c:590 (
>> sofia/internal/1005 at 10.10.10.16) State REPORTING going to sleep
>> 2011-06-12 10:57:57.637437 [DEBUG] switch_core_state_machine.c:327 (
>> sofia/internal/1005 at 10.10.10.16) State Change CS_REPORTING -> CS_DESTROY
>> 2011-06-12 10:57:57.637437 [DEBUG] switch_core_session.c:1021 Send signal
>> sofia/internal/1005 at 10.10.10.16 [BREAK]
>> 2011-06-12 10:57:57.637437 [DEBUG] switch_core_session.c:1164 Session 24 (
>> sofia/internal/1005 at 10.10.10.16) Locked, Waiting on external entities
>> 2011-06-12 10:57:57.637437 [NOTICE] switch_core_session.c:1182 Session 24
>> (sofia/internal/1005 at 10.10.10.16) Ended
>> 2011-06-12 10:57:57.637437 [NOTICE] switch_core_session.c:1184 Close
>> Channel sofia/internal/1005 at 10.10.10.16 [CS_DESTROY]
>> 2011-06-12 10:57:57.637437 [DEBUG] switch_core_state_machine.c:428 (
>> sofia/internal/1005 at 10.10.10.16) Running State Change CS_DESTROY
>> 2011-06-12 10:57:57.637437 [DEBUG] switch_core_state_machine.c:439 (
>> sofia/internal/1005 at 10.10.10.16) State DESTROY
>> 2011-06-12 10:57:57.637437 [DEBUG] mod_sofia.c:341
>> sofia/internal/1005 at 10.10.10.16 SOFIA DESTROY
>> 2011-06-12 10:57:57.637437 [DEBUG] switch_core_state_machine.c:60
>> sofia/internal/1005 at 10.10.10.16 Standard DESTROY
>> 2011-06-12 10:57:57.637437 [DEBUG] switch_core_state_machine.c:439 (
>> sofia/internal/1005 at 10.10.10.16) State DESTROY going to sleep
>> 2011-06-12 10:57:57.720295 [DEBUG] switch_core_session.c:641 Send signal
>> sofia/internal/1234567890 at 10.10.10.16 [BREAK]
>> 2011-06-12 10:57:57.721283 [NOTICE] switch_core_state_machine.c:185
>> sofia/internal/1234567890 at 10.10.10.16 has executed the last dialplan
>> instruction, hanging up.
>> 2011-06-12 10:57:57.721283 [NOTICE] switch_core_state_machine.c:187 Hangup
>> sofia/internal/1234567890 at 10.10.10.16 [CS_EXECUTE] [NORMAL_CLEARING]
>> 2011-06-12 10:57:57.721283 [DEBUG] switch_channel.c:2102 Send signal
>> sofia/internal/1234567890 at 10.10.10.16 [KILL]
>> 2011-06-12 10:57:57.722210 [DEBUG] switch_core_session.c:1021 Send signal
>> sofia/internal/1234567890 at 10.10.10.16 [BREAK]
>> 2011-06-12 10:57:57.722210 [DEBUG] switch_core_state_machine.c:348 (
>> sofia/internal/1234567890 at 10.10.10.16) State EXECUTE going to sleep
>> 2011-06-12 10:57:57.722210 [DEBUG] switch_core_state_machine.c:314 (
>> sofia/internal/1234567890 at 10.10.10.16) Running State Change CS_HANGUP
>> 2011-06-12 10:57:57.722210 [DEBUG] switch_core_state_machine.c:499 (
>> sofia/internal/1234567890 at 10.10.10.16) State HANGUP
>> 2011-06-12 10:57:57.722210 [DEBUG] mod_sofia.c:414 Channel
>> sofia/internal/1234567890 at 10.10.10.16 hanging up, cause: NORMAL_CLEARING
>> 2011-06-12 10:57:57.722210 [DEBUG] mod_sofia.c:476 Responding to INVITE
>> with: 480
>> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:46
>> sofia/internal/1234567890 at 10.10.10.16 Standard HANGUP, cause:
>> NORMAL_CLEARING
>> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:499 (
>> sofia/internal/1234567890 at 10.10.10.16) State HANGUP going to sleep
>> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:333 (
>> sofia/internal/1234567890 at 10.10.10.16) State Change CS_HANGUP ->
>> CS_REPORTING
>> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_session.c:1021 Send signal
>> sofia/internal/1234567890 at 10.10.10.16 [BREAK]
>> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:314 (
>> sofia/internal/1234567890 at 10.10.10.16) Running State Change CS_REPORTING
>> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:590 (
>> sofia/internal/1234567890 at 10.10.10.16) State REPORTING
>> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:53
>> sofia/internal/1234567890 at 10.10.10.16 Standard REPORTING, cause:
>> NORMAL_CLEARING
>> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:590 (
>> sofia/internal/1234567890 at 10.10.10.16) State REPORTING going to sleep
>> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:327 (
>> sofia/internal/1234567890 at 10.10.10.16) State Change CS_REPORTING ->
>> CS_DESTROY
>> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_session.c:1021 Send signal
>> sofia/internal/1234567890 at 10.10.10.16 [BREAK]
>> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_session.c:1164 Session 25 (
>> sofia/internal/1234567890 at 10.10.10.16) Locked, Waiting on external
>> entities
>> 2011-06-12 10:57:57.723321 [NOTICE] switch_core_session.c:1182 Session 25
>> (sofia/internal/1234567890 at 10.10.10.16) Ended
>> 2011-06-12 10:57:57.723321 [NOTICE] switch_core_session.c:1184 Close
>> Channel sofia/internal/1234567890 at 10.10.10.16 [CS_DESTROY]
>> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:428 (
>> sofia/internal/1234567890 at 10.10.10.16) Running State Change CS_DESTROY
>> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:439 (
>> sofia/internal/1234567890 at 10.10.10.16) State DESTROY
>> 2011-06-12 10:57:57.723321 [DEBUG] mod_sofia.c:341
>> sofia/internal/1234567890 at 10.10.10.16 SOFIA DESTROY
>> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:60
>> sofia/internal/1234567890 at 10.10.10.16 Standard DESTROY
>> 2011-06-12 10:57:57.723321 [DEBUG] switch_core_state_machine.c:439 (
>> sofia/internal/1234567890 at 10.10.10.16) State DESTROY going to sleep
>>
>> ##################################################################################
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Jun 16 00:03:02 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 15 Jun 2011 13:03:02 -0700
Subject: [Freeswitch-users] using fs api to originate a call and record
	it
In-Reply-To: <4DF77E9A.6010701@499x.com>
References: <4DF278F9.6000201@499x.com>
	<BANLkTik6Wh7vccJrCx7rUoKSJyYcAzzr3g@mail.gmail.com>
	<4DF778DD.7060805@499x.com> <4DF77E9A.6010701@499x.com>
Message-ID: <BANLkTimtgaj81mSzzHPPfgbQwY7gp_yFpg@mail.gmail.com>

You can use ignore_early_media=true if your scenario allows it.
-MC

On Tue, Jun 14, 2011 at 8:30 AM, <wes-fs at 499x.com> wrote:

>  based on the documentation here, it looks like I might be able to execute
> a script, instead of directly doing the record command.  Then the script
> could maybe set the parameters I need?
>
> http://wiki.freeswitch.org/wiki/Mod_commands#originate
>
>
>
>
> On 6/14/2011 10:06 AM, wes-fs at 499x.com wrote:
>
> This did not work, I still heard the ringing.
>
> On 6/10/2011 5:30 PM, Michael Collins wrote:
>
> Try this:
>
>  originate {
> media_bug_answer_req=true}sofia/sipinterface_1/912625551212 at 141.xxx.xxx.xxx
>  &record(/tmp/myrecording.wav)
>
>  Let us know if it works.
> -MC
>
> On Fri, Jun 10, 2011 at 1:05 PM, <wes-fs at 499x.com> wrote:
>
>> I'm extremely new to freeswitch, and anything like it, so please forgive
>> me!
>>
>> I'd like to be able to use the API to originate a call and record it.
>> So far, I have the following figured out:
>>
>> originate sofia/sipinterface_1/912625551212 at 141.xxx.xxx.xxx
>> &record(/tmp/myrecording.wav)
>> And it dials out to an external number and successfully records the
>> conversation.
>>
>> But, it also records the ringing.  When I was trying this through a
>> dialplan, I found an option:
>> <action application="set" data="media_bug_answer_req=true"/>
>> which delayed the recording until the call was answered.
>>
>> Is there a way to do the same thing through the API?
>>
>> Thanks!
>> Wes
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Jun 16 00:08:31 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 15 Jun 2011 13:08:31 -0700
Subject: [Freeswitch-users] failure to start a session from within lua
 for outbound call
In-Reply-To: <DE04BDDC-761E-484C-938A-A7FE598F3988@iamfine.com>
References: <DE04BDDC-761E-484C-938A-A7FE598F3988@iamfine.com>
Message-ID: <BANLkTik1=M30tBA0uqcoifTws9JGyfPfrA@mail.gmail.com>

Your dialstring is invalid. You need to format it properly:
sofia/gateway/my_gw/1000
sofia/internal/1000 at host
user/1000

It all depends on exactly what you're doing. Use this wiki page for lots of
cool examples:
http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings

-MC

On Tue, Jun 14, 2011 at 1:41 PM, paul at iamfine.com <paul at iamfine.com> wrote:

> Having problems getting a session created from a lua script - its the
> latest build on a centos system
>
> this is the lua code
>
> -----------------------
> freeswitch.console_log("info", "Lua in da house!!!\n");
>
> local session = freeswitch.Session("sofia/00.00.127.61/1000");
> session:execute("playback", "/sr8k.wav");
> session:hangup();
> -----------------------
>
> ***** please note i changed the ip address to protect its identity *****
> **********
>
>
>
>
> please help - i am tearing my hair out on this one
>
> Paul
>
>
> +OK
>
> freeswitch at internal> 2011-06-14 20:31:01.476406 [INFO] switch_cpp.cpp:1197
> Lua in da house!!!
> 2011-06-14 20:31:01.476406 [DEBUG] switch_ivr_originate.c:1869 Parsing
> global variables
> 2011-06-14 20:31:01.476406 [NOTICE] switch_channel.c:833 New Channel
> sofia/internal/1000 [c9c5d8ec-af7f-4153-a844-5000983170d5]
> 2011-06-14 20:31:01.476406 [DEBUG] mod_sofia.c:4392 (sofia/internal/1000)
> State Change CS_NEW -> CS_INIT
> 2011-06-14 20:31:01.476406 [DEBUG] switch_core_session.c:1118 Send signal
> sofia/internal/1000 [BREAK]
> 2011-06-14 20:31:01.476406 [DEBUG] switch_core_state_machine.c:325
> (sofia/internal/1000) Running State Change CS_INIT
> 2011-06-14 20:31:01.476406 [DEBUG] switch_core_state_machine.c:361
> (sofia/internal/1000) State INIT
> 2011-06-14 20:31:01.476406 [DEBUG] mod_sofia.c:85 sofia/internal/1000 SOFIA
> INIT
> 2011-06-14 20:31:01.476406 [DEBUG] mod_sofia.c:125 (sofia/internal/1000)
> State Change CS_INIT -> CS_ROUTING
> 2011-06-14 20:31:01.476406 [DEBUG] switch_core_session.c:1118 Send signal
> sofia/internal/1000 [BREAK]
> 2011-06-14 20:31:01.476406 [DEBUG] switch_core_state_machine.c:361
> (sofia/internal/1000) State INIT going to sleep
> 2011-06-14 20:31:01.476406 [DEBUG] switch_core_state_machine.c:325
> (sofia/internal/1000) Running State Change CS_ROUTING
> 2011-06-14 20:31:01.476406 [DEBUG] switch_channel.c:1736
> (sofia/internal/1000) Callstate Change DOWN -> RINGING
> 2011-06-14 20:31:01.476406 [DEBUG] switch_core_state_machine.c:364
> (sofia/internal/1000) State ROUTING
> 2011-06-14 20:31:01.476406 [DEBUG] mod_sofia.c:148 sofia/internal/1000
> SOFIA ROUTING
> 2011-06-14 20:31:01.476406 [DEBUG] switch_ivr_originate.c:66
> (sofia/internal/1000) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2011-06-14 20:31:01.476406 [DEBUG] switch_core_session.c:1118 Send signal
> sofia/internal/1000 [BREAK]
> 2011-06-14 20:31:01.476406 [DEBUG] switch_core_state_machine.c:364
> (sofia/internal/1000) State ROUTING going to sleep
> 2011-06-14 20:31:01.476406 [DEBUG] switch_core_state_machine.c:325
> (sofia/internal/1000) Running State Change CS_CONSUME_MEDIA
> 2011-06-14 20:31:01.476406 [DEBUG] switch_core_state_machine.c:383
> (sofia/internal/1000) State CONSUME_MEDIA
> 2011-06-14 20:31:01.476406 [DEBUG] switch_core_state_machine.c:383
> (sofia/internal/1000) State CONSUME_MEDIA going to sleep
> 2011-06-14 20:31:01.476406 [DEBUG] sofia.c:4800 Channel sofia/internal/1000
> entering state [calling][0]
> 2011-06-14 20:31:01.638509 [DEBUG] sofia.c:4800 Channel sofia/internal/1000
> entering state [terminated][503]
> 2011-06-14 20:31:01.638509 [DEBUG] switch_channel.c:2641
> (sofia/internal/1000) Callstate Change RINGING -> HANGUP
> 2011-06-14 20:31:01.638509 [NOTICE] sofia.c:5522 Hangup sofia/internal/1000
> [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
> 2011-06-14 20:31:01.638509 [DEBUG] switch_ivr_originate.c:3308 Originate
> Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE]
> 2011-06-14 20:31:01.638509 [ERR] switch_cpp.cpp:655 session is not
> initalized
> 2011-06-14 20:31:01.638509 [ERR] switch_cpp.cpp:617 session is not
> initalized
> 2011-06-14 20:31:01.638509 [DEBUG] switch_channel.c:2657 Send signal
> sofia/internal/1000 [KILL]
> 2011-06-14 20:31:01.638509 [DEBUG] switch_core_session.c:1118 Send signal
> sofia/internal/1000 [BREAK]
> 2011-06-14 20:31:01.638509 [DEBUG] switch_core_state_machine.c:325
> (sofia/internal/1000) Running State Change CS_HANGUP
> 2011-06-14 20:31:01.638509 [DEBUG] switch_core_state_machine.c:565
> (sofia/internal/1000) State HANGUP
> 2011-06-14 20:31:01.638509 [DEBUG] mod_sofia.c:452 sofia/internal/1000
> Overriding SIP cause 503 with 503 from the other leg
> 2011-06-14 20:31:01.638509 [DEBUG] mod_sofia.c:458 Channel
> sofia/internal/1000 hanging up, cause: NORMAL_TEMPORARY_FAILURE
> 2011-06-14 20:31:01.638509 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/1000 Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE
> 2011-06-14 20:31:01.638509 [DEBUG] switch_core_state_machine.c:565
> (sofia/internal/1000) State HANGUP going to sleep
> 2011-06-14 20:31:01.638509 [DEBUG] switch_core_state_machine.c:356
> (sofia/internal/1000) State Change CS_HANGUP -> CS_REPORTING
> 2011-06-14 20:31:01.638509 [DEBUG] switch_core_session.c:1118 Send signal
> sofia/internal/1000 [BREAK]
> 2011-06-14 20:31:01.638509 [DEBUG] switch_core_state_machine.c:325
> (sofia/internal/1000) Running State Change CS_REPORTING
> 2011-06-14 20:31:01.638509 [DEBUG] switch_core_state_machine.c:625
> (sofia/internal/1000) State REPORTING
> 2011-06-14 20:31:02.631557 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/1000 Standard REPORTING, cause: NORMAL_TEMPORARY_FAILURE
> 2011-06-14 20:31:02.631557 [DEBUG] switch_core_state_machine.c:625
> (sofia/internal/1000) State REPORTING going to sleep
> 2011-06-14 20:31:02.631557 [DEBUG] switch_core_state_machine.c:350
> (sofia/internal/1000) State Change CS_REPORTING -> CS_DESTROY
> 2011-06-14 20:31:02.631557 [DEBUG] switch_core_session.c:1118 Send signal
> sofia/internal/1000 [BREAK]
> 2011-06-14 20:31:02.631557 [DEBUG] switch_core_session.c:1290 Session 8
> (sofia/internal/1000) Locked, Waiting on external entities
> 2011-06-14 20:31:02.631557 [NOTICE] switch_core_session.c:1308 Session 8
> (sofia/internal/1000) Ended
> 2011-06-14 20:31:02.631557 [NOTICE] switch_core_session.c:1310 Close
> Channel sofia/internal/1000 [CS_DESTROY]
> 2011-06-14 20:31:02.631557 [DEBUG] switch_core_state_machine.c:454
> (sofia/internal/1000) Callstate Change HANGUP -> DOWN
> 2011-06-14 20:31:02.631557 [DEBUG] switch_core_state_machine.c:457
> (sofia/internal/1000) Running State Change CS_DESTROY
> 2011-06-14 20:31:02.631557 [DEBUG] switch_core_state_machine.c:467
> (sofia/internal/1000) State DESTROY
> 2011-06-14 20:31:02.631557 [DEBUG] mod_sofia.c:363 sofia/internal/1000
> SOFIA DESTROY
> 2011-06-14 20:31:02.631557 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/1000 Standard DESTROY
> 2011-06-14 20:31:02.631557 [DEBUG] switch_core_state_machine.c:467
> (sofia/internal/1000) State DESTROY going to sleep
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gmaruzz at gmail.com  Thu Jun 16 00:21:46 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 15 Jun 2011 22:21:46 +0200
Subject: [Freeswitch-users] unixODBC MySQL mysqld-5.0.77 - error
 messages with mod_callcenter
In-Reply-To: <5003D7D3E06F514E8C682F18D223265C04D3B36C29@AZWSMS03.azwarranty.int>
References: <5003D7D3E06F514E8C682F18D223265C04D3B36C29@AZWSMS03.azwarranty.int>
Message-ID: <BANLkTim3oqceLgziN75cVJAY5c78aoMBSA@mail.gmail.com>

Stefan,
your mail is unreadable.
Send plain old pure ASCII, not fancy cryptostuff :)
-giovanni

On Wed, Jun 15, 2011 at 5:14 PM, Weigel, Stefan <
Stefan.Weigel at allianz-warranty.com> wrote:

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



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From msc at freeswitch.org  Thu Jun 16 01:09:37 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 15 Jun 2011 14:09:37 -0700
Subject: [Freeswitch-users] in bypass_media mode,
 2833 payload type modified by freeswitch!
In-Reply-To: <BANLkTikfoYbL-atXM78W7BTLe3LNUheF9A@mail.gmail.com>
References: <BANLkTi=HpPE7q4Kbrkwe-KJxj69SYJtx0Q@mail.gmail.com>
	<4A5154F6-90DE-4B40-9F24-E3E0D0824D02@ipeva.fr>
	<BANLkTikfoYbL-atXM78W7BTLe3LNUheF9A@mail.gmail.com>
Message-ID: <BANLkTikMc=Pdwx6W85JBHZ4AJxpKhdhz5A@mail.gmail.com>

QQ,

I don't see that FreeSWITCH is doing anything incorrect here. According to
RFC3264, the offerer (FreeSWITCH) sends an SDP and in the case of RTP, the
answerer (GRSIP Gateway) must use the payload type offered, even if the
answerer uses a different payload type when it sends a telephone-event.

http://tools.ietf.org/html/rfc3264#section-6.1
Specifically near the end:

"In the case of RTP, it MUST use the payload type numbers

   from the offer, even if they differ from those in the answer."


Technically, FreeSWITCH isn't "changing" anything anyway. The originator of
the call (Vox Callcontrol) is the one who chose PT 101 in the INVITE that it
sent to FreeSWITCH. FS is just passing that along without modifying it.

I think you need to contact the people running the gateway and make sure
that they understand that they are not following RFC3264 if they're
rejecting telephone-events in PT 101 simply because they prefer to send in
PT 110. Also, if your Vox Callcontrol client has any configuration options
then maybe you can tell it to use PT 110 for RFC2833 DTMFs. It is a bit of a
workaround but this is SIP, so no one is expecting perfection. :)

-MC

On Wed, Jun 15, 2011 at 12:23 AM, qingquan luo <qingquan at globalroam.com>wrote:

> Hi David,
>      Thank you for reminder,   I resend this mail in mail list, Because I
> can not reply preview mail( I make a mistake to subscribe this mail list in
> Daily Digest mode.)
>      So I need resend this mail for require help.
>
> For this issue, I has paste sip debug output in:
>
> information with "sofia global siptrace on"
> http://pastebin.freeswitch.org/16496
>
>
>  Thanks
>
>
>
> On Wed, Jun 15, 2011 at 3:15 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>
>> You have a mail issue. Check your spam...
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 15/06/2011 ? 03:55, qingquan luo a ?crit :
>>
>> Hi All,
>>
>>   I use freeswitch to bridge one incoming call to other target phone
>> number.
>>   I use bypass_media mode.
>>   So the rtp is not go through the freeswitch,
>>   By I notice that When LegB reply 183 or 200 SDP message. Freeswitch
>> change it 2833 telephone-event payload type. and forwarding the message to
>> caller. This make the caller use wrong dtmf payload type to send dtmf
>>
>>   What wrong with that?  How can fix it?
>>   Any information or help is welcome.
>>
>> information with "sofia global siptrace on"
>> http://pastebin.freeswitch.org/16496
>>
>> Thanks
>>
>>
>> Best Regards
>>
>>
>> Qingquan
>>
>> --
>> Using Gmail? Please read this important notice:
>> http://www.fsf.org/campaigns/jstrap/gmail?40922.
>>  _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Using Gmail? Please read this important notice:
> http://www.fsf.org/campaigns/jstrap/gmail?40922.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kris at kriskinc.com  Thu Jun 16 01:58:47 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 15 Jun 2011 17:58:47 -0400
Subject: [Freeswitch-users] in bypass_media mode,
 2833 payload type modified by freeswitch!
In-Reply-To: <BANLkTikMc=Pdwx6W85JBHZ4AJxpKhdhz5A@mail.gmail.com>
References: <BANLkTi=HpPE7q4Kbrkwe-KJxj69SYJtx0Q@mail.gmail.com>
	<4A5154F6-90DE-4B40-9F24-E3E0D0824D02@ipeva.fr>
	<BANLkTikfoYbL-atXM78W7BTLe3LNUheF9A@mail.gmail.com>
	<BANLkTikMc=Pdwx6W85JBHZ4AJxpKhdhz5A@mail.gmail.com>
Message-ID: <BANLkTikSr75c8cucBZcL+bkkTisuy6X1zw@mail.gmail.com>

Michael,

  This isn't an SDP offer/answer issue with FreeSWITCH.  Even if there
were underlying offer/answer issues between the endpoints FreeSWITCH
is in bypass_media mode and it shouldn't care about the SDP because by
definition (bypass_media) the SDP is "pass through" between the two
remote endpoints.

  The specific issue lies in the 183 w/ SDP at line 207.  It has
rfc2833 payload type 110.  When FreeSWITCH forwards this to the other
end (line 276) the SDP is untouched (expected with bypass_media)
EXCEPT the rfc2833 payload type has been changed to 101 (unexpected
with bypass_media).  It does this again with the 200 OK.  All other
SDP params are passed through - media address, port, codec, goofy
session name, and even the SER rtpproxy attributes.  I'll admit the
remote end/proxy is doing some strange stuff - 100, 101, 183, then
180, etc but this does look like a strange bypass_media bug in
FreeSWITCH.  The RFC2833 payload type should be forwarded between the
two remote endpoints without being modified by FreeSWITCH - just like
all of the other SDP parameters (or any part of the SIP body, for that
matter).

On Wed, Jun 15, 2011 at 5:09 PM, Michael Collins <msc at freeswitch.org> wrote:
> QQ,
> I don't see that FreeSWITCH is doing anything incorrect here. According to
> RFC3264, the offerer (FreeSWITCH) sends an SDP and in the case of RTP, the
> answerer (GRSIP Gateway) must use the payload type offered, even if the
> answerer uses a different payload type when it sends a telephone-event.
> http://tools.ietf.org/html/rfc3264#section-6.1
> Specifically near the end:
> "In the case of RTP, it MUST use the payload type numbers
>
>    from the offer, even if they differ from those in the answer."
>
> Technically, FreeSWITCH isn't "changing" anything anyway. The originator of
> the call (Vox Callcontrol) is the one who chose PT 101 in the INVITE that it
> sent to FreeSWITCH. FS is just passing that along without modifying it.
> I think you need to contact the people running the gateway and make sure
> that they understand that they are not following RFC3264 if they're
> rejecting telephone-events in PT 101 simply because they prefer to send in
> PT 110. Also, if your Vox Callcontrol client has any configuration options
> then maybe you can tell it to use PT 110 for RFC2833 DTMFs. It is a bit of a
> workaround but this is SIP, so no one is expecting perfection. :)
> -MC

-- 
Kristian Kielhofner


From wes-fs at 499x.com  Thu Jun 16 02:04:36 2011
From: wes-fs at 499x.com (Wes)
Date: Wed, 15 Jun 2011 17:04:36 -0500
Subject: [Freeswitch-users] using fs api to originate a call and record
 it
In-Reply-To: <BANLkTimtgaj81mSzzHPPfgbQwY7gp_yFpg@mail.gmail.com>
References: <4DF278F9.6000201@499x.com>	<BANLkTik6Wh7vccJrCx7rUoKSJyYcAzzr3g@mail.gmail.com>	<4DF778DD.7060805@499x.com>
	<4DF77E9A.6010701@499x.com>
	<BANLkTimtgaj81mSzzHPPfgbQwY7gp_yFpg@mail.gmail.com>
Message-ID: <4DF92C74.6010607@499x.com>

I've successfully transitioned to using a lua script instead of using 
the &record command directly:

something like this:

in the api:
originate sofia/sipinterface_1/8839 at 1xxx.xx.xx &lua(hello.lua)

and in the hello.lua script:

session:answer();
session:recordFile("/tmp/myrecording.wav");
session:hangup();

this is much more flexible than doing:

originate sofia/sipinterface_1/8839 at 141.106.29.6 
&record('/tmp/myrecording.wav')

On 6/15/2011 3:03 PM, Michael Collins wrote:
> You can use ignore_early_media=true if your scenario allows it.
> -MC
>
> On Tue, Jun 14, 2011 at 8:30 AM, <wes-fs at 499x.com 
> <mailto:wes-fs at 499x.com>> wrote:
>
>     based on the documentation here, it looks like I might be able to
>     execute a script, instead of directly doing the record command. 
>     Then the script could maybe set the parameters I need?
>
>     http://wiki.freeswitch.org/wiki/Mod_commands#originate
>
>
>
>
>     On 6/14/2011 10:06 AM, wes-fs at 499x.com <mailto:wes-fs at 499x.com>
>     wrote:
>>     This did not work, I still heard the ringing.
>>
>>     On 6/10/2011 5:30 PM, Michael Collins wrote:
>>>     Try this:
>>>
>>>     originate
>>>     {media_bug_answer_req=true}sofia/sipinterface_1/912625551212 at 141.xxx.xxx.xxx
>>>     <mailto:media_bug_answer_req=true%7Dsofia/sipinterface_1/912625551212 at 141.xxx.xxx.xxx>
>>>     &record(/tmp/myrecording.wav)
>>>
>>>     Let us know if it works.
>>>     -MC
>>>
>>>     On Fri, Jun 10, 2011 at 1:05 PM, <wes-fs at 499x.com
>>>     <mailto:wes-fs at 499x.com>> wrote:
>>>
>>>         I'm extremely new to freeswitch, and anything like it, so
>>>         please forgive me!
>>>
>>>         I'd like to be able to use the API to originate a call and
>>>         record it.
>>>         So far, I have the following figured out:
>>>
>>>         originate sofia/sipinterface_1/912625551212 at 141.xxx.xxx.xxx
>>>         <mailto:sofia/sipinterface_1/912625551212 at 141.xxx.xxx.xxx>
>>>         &record(/tmp/myrecording.wav)
>>>         And it dials out to an external number and successfully
>>>         records the
>>>         conversation.
>>>
>>>         But, it also records the ringing.  When I was trying this
>>>         through a
>>>         dialplan, I found an option:
>>>         <action application="set" data="media_bug_answer_req=true"/>
>>>         which delayed the recording until the call was answered.
>>>
>>>         Is there a way to do the same thing through the API?
>>>
>>>         Thanks!
>>>         Wes
>>>
>>>         _______________________________________________
>>>         FreeSWITCH-users mailing list
>>>         FreeSWITCH-users at lists.freeswitch.org
>>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>         http://www.freeswitch.org
>>>
>>>
>>>
>>>     _______________________________________________
>>>     FreeSWITCH-users mailing list
>>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From kris at kriskinc.com  Thu Jun 16 02:06:13 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 15 Jun 2011 18:06:13 -0400
Subject: [Freeswitch-users] 183 after 180
In-Reply-To: <9AD5D885-6663-4C1D-B3FA-7CFC1C49F68F@teliax.com>
References: <9AD5D885-6663-4C1D-B3FA-7CFC1C49F68F@teliax.com>
Message-ID: <BANLkTin+fvcVSfiaAXb+vCETTV387UL+XA@mail.gmail.com>

David,

  While the solution you've proposed below is the best in terms of
media handling I'm not sure how well it will work in practice.

  If you intercept the 180 and provide 183 w/ media the calling
endpoint will get media (ringback) from the IP address of your FS
machine (no problem).  However, when the 183/200 w/ media comes from
the remote end it is going to contain a different IP address and port
if you (somehow) re-enable bypass_media.  Even if FreeSWITCH could be
coerced into doing this I'm not sure that all endpoints will react
well to this change in connection information from your 183 to the
remote 183/200.  Technically speaking (RFC) there's nothing wrong with
it but we all know that doesn't mean there won't be issues doing this
with the various endpoints in the field.

  You /may/ get stuck anchoring the media for the entire call in these
cases where your upstream returns a 180.  Other than that I'd be
interested to see how you get along with this...

On Wed, Jun 15, 2011 at 12:24 AM, David Aldworth <daldworth at teliax.com> wrote:
> I am trying to find the best solution to generate ring on 180 but bypass on 183. We are in the middle, customer -> FS -> pstn. We bypass_media=true for all calls. However, sometimes we get a 180 from the PSTN and FS relays that to the customer, but the customer is not generating ring so the calling party hears dead air. So, the solution would be to anchor media and generate ring when we get a 180, but once the 183 or 200 hits, bypass starts. I suppose we could do an expression match on the 180, ignore early media, generate ring, and bypass media after bridge. Just wondering if there was a better alternative solution.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner


From msc at freeswitch.org  Thu Jun 16 02:11:05 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 15 Jun 2011 15:11:05 -0700
Subject: [Freeswitch-users] in bypass_media mode,
 2833 payload type modified by freeswitch!
In-Reply-To: <BANLkTikSr75c8cucBZcL+bkkTisuy6X1zw@mail.gmail.com>
References: <BANLkTi=HpPE7q4Kbrkwe-KJxj69SYJtx0Q@mail.gmail.com>
	<4A5154F6-90DE-4B40-9F24-E3E0D0824D02@ipeva.fr>
	<BANLkTikfoYbL-atXM78W7BTLe3LNUheF9A@mail.gmail.com>
	<BANLkTikMc=Pdwx6W85JBHZ4AJxpKhdhz5A@mail.gmail.com>
	<BANLkTikSr75c8cucBZcL+bkkTisuy6X1zw@mail.gmail.com>
Message-ID: <BANLkTi=F2U+VPJGdRWjot6wTktZrF3ZCAg@mail.gmail.com>

Ah, quite right. I'd file this in Jira. Hopefully it's not buried deep in
the Sofia stack...

-MC

On Wed, Jun 15, 2011 at 2:58 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> Michael,
>
>  This isn't an SDP offer/answer issue with FreeSWITCH.  Even if there
> were underlying offer/answer issues between the endpoints FreeSWITCH
> is in bypass_media mode and it shouldn't care about the SDP because by
> definition (bypass_media) the SDP is "pass through" between the two
> remote endpoints.
>
>  The specific issue lies in the 183 w/ SDP at line 207.  It has
> rfc2833 payload type 110.  When FreeSWITCH forwards this to the other
> end (line 276) the SDP is untouched (expected with bypass_media)
> EXCEPT the rfc2833 payload type has been changed to 101 (unexpected
> with bypass_media).  It does this again with the 200 OK.  All other
> SDP params are passed through - media address, port, codec, goofy
> session name, and even the SER rtpproxy attributes.  I'll admit the
> remote end/proxy is doing some strange stuff - 100, 101, 183, then
> 180, etc but this does look like a strange bypass_media bug in
> FreeSWITCH.  The RFC2833 payload type should be forwarded between the
> two remote endpoints without being modified by FreeSWITCH - just like
> all of the other SDP parameters (or any part of the SIP body, for that
> matter).
>
> On Wed, Jun 15, 2011 at 5:09 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> > QQ,
> > I don't see that FreeSWITCH is doing anything incorrect here. According
> to
> > RFC3264, the offerer (FreeSWITCH) sends an SDP and in the case of RTP,
> the
> > answerer (GRSIP Gateway) must use the payload type offered, even if the
> > answerer uses a different payload type when it sends a telephone-event.
> > http://tools.ietf.org/html/rfc3264#section-6.1
> > Specifically near the end:
> > "In the case of RTP, it MUST use the payload type numbers
> >
> >    from the offer, even if they differ from those in the answer."
> >
> > Technically, FreeSWITCH isn't "changing" anything anyway. The originator
> of
> > the call (Vox Callcontrol) is the one who chose PT 101 in the INVITE that
> it
> > sent to FreeSWITCH. FS is just passing that along without modifying it.
> > I think you need to contact the people running the gateway and make sure
> > that they understand that they are not following RFC3264 if they're
> > rejecting telephone-events in PT 101 simply because they prefer to send
> in
> > PT 110. Also, if your Vox Callcontrol client has any configuration
> options
> > then maybe you can tell it to use PT 110 for RFC2833 DTMFs. It is a bit
> of a
> > workaround but this is SIP, so no one is expecting perfection. :)
> > -MC
>
> --
> Kristian Kielhofner
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From daldworth at teliax.com  Thu Jun 16 02:35:03 2011
From: daldworth at teliax.com (David Aldworth)
Date: Wed, 15 Jun 2011 16:35:03 -0600
Subject: [Freeswitch-users] 183 after 180
In-Reply-To: <BANLkTin+fvcVSfiaAXb+vCETTV387UL+XA@mail.gmail.com>
References: <9AD5D885-6663-4C1D-B3FA-7CFC1C49F68F@teliax.com>
	<BANLkTin+fvcVSfiaAXb+vCETTV387UL+XA@mail.gmail.com>
Message-ID: <EC6AC96E-7FAC-4BB4-9FF9-1711CB33094C@teliax.com>

Your comments about different endpoints reacting different is my concern exactly. That said, we came up with a fairly simple solution (below) that, thus far, seems alright. However, not all scenarios have been fully test, such as what happens if we get an early media operator intercept with the 183, what happens if the endpoint rejects the re-invite, etc.

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


On Jun 15, 2011, at 4:06 PM, Kristian Kielhofner wrote:

> David,
> 
>  While the solution you've proposed below is the best in terms of
> media handling I'm not sure how well it will work in practice.
> 
>  If you intercept the 180 and provide 183 w/ media the calling
> endpoint will get media (ringback) from the IP address of your FS
> machine (no problem).  However, when the 183/200 w/ media comes from
> the remote end it is going to contain a different IP address and port
> if you (somehow) re-enable bypass_media.  Even if FreeSWITCH could be
> coerced into doing this I'm not sure that all endpoints will react
> well to this change in connection information from your 183 to the
> remote 183/200.  Technically speaking (RFC) there's nothing wrong with
> it but we all know that doesn't mean there won't be issues doing this
> with the various endpoints in the field.
> 
>  You /may/ get stuck anchoring the media for the entire call in these
> cases where your upstream returns a 180.  Other than that I'd be
> interested to see how you get along with this...
> 
> On Wed, Jun 15, 2011 at 12:24 AM, David Aldworth <daldworth at teliax.com> wrote:
>> I am trying to find the best solution to generate ring on 180 but bypass on 183. We are in the middle, customer -> FS -> pstn. We bypass_media=true for all calls. However, sometimes we get a 180 from the PSTN and FS relays that to the customer, but the customer is not generating ring so the calling party hears dead air. So, the solution would be to anchor media and generate ring when we get a 180, but once the 183 or 200 hits, bypass starts. I suppose we could do an expression match on the 180, ignore early media, generate ring, and bypass media after bridge. Just wondering if there was a better alternative solution.
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> -- 
> Kristian Kielhofner
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From avi at avimarcus.net  Thu Jun 16 02:44:36 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 16 Jun 2011 01:44:36 +0300
Subject: [Freeswitch-users] Presence SQL Error
Message-ID: <BANLkTi=Gk4DT6skr2Kz2QwEjPmvRSy7vPw@mail.gmail.com>

This is a pretty fresh install of FS on centos, using sqlite for the DB.
I do have multiple domains set up.

There's an SQL error for the presence requests, but FS isn't telling me much
info. What can I do about this?
The only times I've used presence, it basically just worked.

http://pastebin.freeswitch.org/16518
 <http://pastebin.freeswitch.org/16518>
-Avi
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From anthony.minessale at gmail.com  Thu Jun 16 02:47:03 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 15 Jun 2011 17:47:03 -0500
Subject: [Freeswitch-users] in bypass_media mode,
 2833 payload type modified by freeswitch!
In-Reply-To: <BANLkTi=F2U+VPJGdRWjot6wTktZrF3ZCAg@mail.gmail.com>
References: <BANLkTi=HpPE7q4Kbrkwe-KJxj69SYJtx0Q@mail.gmail.com>
	<4A5154F6-90DE-4B40-9F24-E3E0D0824D02@ipeva.fr>
	<BANLkTikfoYbL-atXM78W7BTLe3LNUheF9A@mail.gmail.com>
	<BANLkTikMc=Pdwx6W85JBHZ4AJxpKhdhz5A@mail.gmail.com>
	<BANLkTikSr75c8cucBZcL+bkkTisuy6X1zw@mail.gmail.com>
	<BANLkTi=F2U+VPJGdRWjot6wTktZrF3ZCAg@mail.gmail.com>
Message-ID: <BANLkTikhE7Mj0nM4J0+NN+zEOfVjaoiOCQ@mail.gmail.com>

However, there is not really any rule anywhere that states that bypass
media will not touch the packets.
All of the packets/sdp are still subject to SIP SOA model so if you
pass a remote sdp into the lib, sofia will parse and modify it.
The idea of bypass media is to bypass the media by advertising the
proper codecs/addrs to arrange point to point media and there is no
assurance that the packets will not be modified.

However,

There is a sofia param called "enable-soa" and a variable
sip_enable_soa for per call basis which can be set to false to disable
SIP SOA from sofia and most likely result in untouched exchange of
SDP.

try this in your dialplan

<action application="set" data="bypass_media=true"/>
<action application="export" data="sip_enable_soa=false"/>




On Wed, Jun 15, 2011 at 5:11 PM, Michael Collins <msc at freeswitch.org> wrote:
> Ah, quite right. I'd file this in Jira. Hopefully it's not buried deep in
> the Sofia stack...
> -MC
>
> On Wed, Jun 15, 2011 at 2:58 PM, Kristian Kielhofner <kris at kriskinc.com>
> wrote:
>>
>> Michael,
>>
>> ?This isn't an SDP offer/answer issue with FreeSWITCH. ?Even if there
>> were underlying offer/answer issues between the endpoints FreeSWITCH
>> is in bypass_media mode and it shouldn't care about the SDP because by
>> definition (bypass_media) the SDP is "pass through" between the two
>> remote endpoints.
>>
>> ?The specific issue lies in the 183 w/ SDP at line 207. ?It has
>> rfc2833 payload type 110. ?When FreeSWITCH forwards this to the other
>> end (line 276) the SDP is untouched (expected with bypass_media)
>> EXCEPT the rfc2833 payload type has been changed to 101 (unexpected
>> with bypass_media). ?It does this again with the 200 OK. ?All other
>> SDP params are passed through - media address, port, codec, goofy
>> session name, and even the SER rtpproxy attributes. ?I'll admit the
>> remote end/proxy is doing some strange stuff - 100, 101, 183, then
>> 180, etc but this does look like a strange bypass_media bug in
>> FreeSWITCH. ?The RFC2833 payload type should be forwarded between the
>> two remote endpoints without being modified by FreeSWITCH - just like
>> all of the other SDP parameters (or any part of the SIP body, for that
>> matter).
>>
>> On Wed, Jun 15, 2011 at 5:09 PM, Michael Collins <msc at freeswitch.org>
>> wrote:
>> > QQ,
>> > I don't see that FreeSWITCH is doing anything incorrect here. According
>> > to
>> > RFC3264, the offerer (FreeSWITCH) sends an SDP and in the case of RTP,
>> > the
>> > answerer (GRSIP Gateway) must use the payload type offered, even if the
>> > answerer uses a different payload type when it sends a telephone-event.
>> > http://tools.ietf.org/html/rfc3264#section-6.1
>> > Specifically near the end:
>> > "In the case of RTP, it MUST use the payload type numbers
>> >
>> > ? ?from the offer, even if they differ from those in the answer."
>> >
>> > Technically, FreeSWITCH isn't "changing" anything anyway. The originator
>> > of
>> > the call (Vox Callcontrol) is the one who chose PT 101 in the INVITE
>> > that it
>> > sent to FreeSWITCH. FS is just passing that along without modifying it.
>> > I think you need to contact the people running the gateway and make sure
>> > that they understand that they are not following RFC3264 if they're
>> > rejecting telephone-events in PT 101 simply because they prefer to send
>> > in
>> > PT 110. Also, if your Vox Callcontrol client has any configuration
>> > options
>> > then maybe you can tell it to use PT 110 for RFC2833 DTMFs. It is a bit
>> > of a
>> > workaround but this is SIP, so no one is expecting perfection. :)
>> > -MC
>>
>> --
>> Kristian Kielhofner
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From avi at avimarcus.net  Thu Jun 16 02:49:00 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 16 Jun 2011 01:49:00 +0300
Subject: [Freeswitch-users] Funky Gateway Confusion?
Message-ID: <BANLkTi=6909x6oK_dYEm+u2tZy4oeyP47A@mail.gmail.com>

When making a call out on a gateway named "xconnect" they ping to see if my
machine is still active.
It uses 1111 in there, which happens to be the highest numbered SIP username
on my system.
Oh, and gw=Spikko001 gets in there too which is a completely unrelated
gateway on my system.

What in the world is going on?

http://pastebin.freeswitch.org/16519

<http://pastebin.freeswitch.org/16519>It doesn't seem to be causing any
problems.. but it's so weird.
kinda old: FreeSWITCH Version 1.0.head (git-0911ed7 2011-03-05 12-49-19
-0500)

-Avi Marcus
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From msc at freeswitch.org  Thu Jun 16 03:29:13 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 15 Jun 2011 16:29:13 -0700
Subject: [Freeswitch-users] in bypass_media mode,
 2833 payload type modified by freeswitch!
In-Reply-To: <BANLkTikhE7Mj0nM4J0+NN+zEOfVjaoiOCQ@mail.gmail.com>
References: <BANLkTi=HpPE7q4Kbrkwe-KJxj69SYJtx0Q@mail.gmail.com>
	<4A5154F6-90DE-4B40-9F24-E3E0D0824D02@ipeva.fr>
	<BANLkTikfoYbL-atXM78W7BTLe3LNUheF9A@mail.gmail.com>
	<BANLkTikMc=Pdwx6W85JBHZ4AJxpKhdhz5A@mail.gmail.com>
	<BANLkTikSr75c8cucBZcL+bkkTisuy6X1zw@mail.gmail.com>
	<BANLkTi=F2U+VPJGdRWjot6wTktZrF3ZCAg@mail.gmail.com>
	<BANLkTikhE7Mj0nM4J0+NN+zEOfVjaoiOCQ@mail.gmail.com>
Message-ID: <BANLkTimLLbr+oEh3m0JY7YRSWMDQzP5ABQ@mail.gmail.com>

And FTR, this *is* on the wiki - I just didn't know about it. :)

http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation
(Last item)

-MC

On Wed, Jun 15, 2011 at 3:47 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> However, there is not really any rule anywhere that states that bypass
> media will not touch the packets.
> All of the packets/sdp are still subject to SIP SOA model so if you
> pass a remote sdp into the lib, sofia will parse and modify it.
> The idea of bypass media is to bypass the media by advertising the
> proper codecs/addrs to arrange point to point media and there is no
> assurance that the packets will not be modified.
>
> However,
>
> There is a sofia param called "enable-soa" and a variable
> sip_enable_soa for per call basis which can be set to false to disable
> SIP SOA from sofia and most likely result in untouched exchange of
> SDP.
>
> try this in your dialplan
>
> <action application="set" data="bypass_media=true"/>
> <action application="export" data="sip_enable_soa=false"/>
>
>
>
>
> On Wed, Jun 15, 2011 at 5:11 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> > Ah, quite right. I'd file this in Jira. Hopefully it's not buried deep in
> > the Sofia stack...
> > -MC
> >
> > On Wed, Jun 15, 2011 at 2:58 PM, Kristian Kielhofner <kris at kriskinc.com>
> > wrote:
> >>
> >> Michael,
> >>
> >>  This isn't an SDP offer/answer issue with FreeSWITCH.  Even if there
> >> were underlying offer/answer issues between the endpoints FreeSWITCH
> >> is in bypass_media mode and it shouldn't care about the SDP because by
> >> definition (bypass_media) the SDP is "pass through" between the two
> >> remote endpoints.
> >>
> >>  The specific issue lies in the 183 w/ SDP at line 207.  It has
> >> rfc2833 payload type 110.  When FreeSWITCH forwards this to the other
> >> end (line 276) the SDP is untouched (expected with bypass_media)
> >> EXCEPT the rfc2833 payload type has been changed to 101 (unexpected
> >> with bypass_media).  It does this again with the 200 OK.  All other
> >> SDP params are passed through - media address, port, codec, goofy
> >> session name, and even the SER rtpproxy attributes.  I'll admit the
> >> remote end/proxy is doing some strange stuff - 100, 101, 183, then
> >> 180, etc but this does look like a strange bypass_media bug in
> >> FreeSWITCH.  The RFC2833 payload type should be forwarded between the
> >> two remote endpoints without being modified by FreeSWITCH - just like
> >> all of the other SDP parameters (or any part of the SIP body, for that
> >> matter).
> >>
> >> On Wed, Jun 15, 2011 at 5:09 PM, Michael Collins <msc at freeswitch.org>
> >> wrote:
> >> > QQ,
> >> > I don't see that FreeSWITCH is doing anything incorrect here.
> According
> >> > to
> >> > RFC3264, the offerer (FreeSWITCH) sends an SDP and in the case of RTP,
> >> > the
> >> > answerer (GRSIP Gateway) must use the payload type offered, even if
> the
> >> > answerer uses a different payload type when it sends a
> telephone-event.
> >> > http://tools.ietf.org/html/rfc3264#section-6.1
> >> > Specifically near the end:
> >> > "In the case of RTP, it MUST use the payload type numbers
> >> >
> >> >    from the offer, even if they differ from those in the answer."
> >> >
> >> > Technically, FreeSWITCH isn't "changing" anything anyway. The
> originator
> >> > of
> >> > the call (Vox Callcontrol) is the one who chose PT 101 in the INVITE
> >> > that it
> >> > sent to FreeSWITCH. FS is just passing that along without modifying
> it.
> >> > I think you need to contact the people running the gateway and make
> sure
> >> > that they understand that they are not following RFC3264 if they're
> >> > rejecting telephone-events in PT 101 simply because they prefer to
> send
> >> > in
> >> > PT 110. Also, if your Vox Callcontrol client has any configuration
> >> > options
> >> > then maybe you can tell it to use PT 110 for RFC2833 DTMFs. It is a
> bit
> >> > of a
> >> > workaround but this is SIP, so no one is expecting perfection. :)
> >> > -MC
> >>
> >> --
> >> Kristian Kielhofner
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From justlikeef at gmail.com  Thu Jun 16 05:55:41 2011
From: justlikeef at gmail.com (Rob Hutton)
Date: Wed, 15 Jun 2011 21:55:41 -0400
Subject: [Freeswitch-users] Trying to get TLS/SRTP working
In-Reply-To: <BANLkTimLoVv7sRw-OOY-th0fC94uj-oYzw@mail.gmail.com>
References: <BANLkTimLoVv7sRw-OOY-th0fC94uj-oYzw@mail.gmail.com>
Message-ID: <201106152155.42232.justlikeef@gmail.com>

What I have been able to figure out is that if I try to enable encryption on 
all of the interfaces defined on an IP, then freeswitch does not listen on the 
TLS port.  But if I enable it on only one of the interfaces, then it works.

On Tuesday 14 June 2011 14:01:20 Rob Hutton wrote:
> Trying to get TLS working.  When sofia loads, I see that it seems to accept
> the parameters:
> 
> http://pastebin.freeswitch.org/16487
> 
> But sofia status shows nothing running on the TLS ports:
>                      Name
> Type                                       Data      State
> ===========================================================================
> ====================== sipinterface_3       profile
> sip:mod_sofia at 192.168.2.25:5080      RUNNING (0)
>            sipinterface_2       profile
> sip:mod_sofia at 192.168.2.25:5070      RUNNING (0)
>              192.168.2.25         alias
> sipinterface_1      ALIASED
>            sipinterface_1       profile
> sip:mod_sofia at 192.168.2.25:5060      RUNNING (0)
>               voicemail_1         alias
> sipinterface_1      ALIASED
> ===========================================================================
> ====================== 3 profiles 2 aliases
> 
> 
> /usr/local/freeswitch/conf/ssl looks like:
> -rw-r--r-- 1 root root 3627 Jun 14 12:55 agent.pem
> -rw-r--r-- 1 root root 1996 Jun 14 13:04 cafile.pem
> 
> 
> Freeswitch is compiled with SSL support:
> 
> ldd freeswitch
>         linux-vdso.so.1 =>  (0x00007fff415ff000)
>         libm.so.6 => /lib64/libm.so.6 (0x00007fab636b8000)
>         libfreeswitch.so.1 => /usr/local/freeswitch/lib/libfreeswitch.so.1
> (0x00007fab632c1000)
>         libuuid.so.1 => /lib64/libuuid.so.1 (0x00007fab630bc000)
>         librt.so.1 => /lib64/librt.so.1 (0x00007fab62eb3000)
>         libcrypt.so.1 => /lib64/libcrypt.so.1 (0x00007fab62c78000)
>         libpthread.so.0 => /lib64/libpthread.so.0 (0x00007fab62a5b000)
>         libssl.so.1.0.0 => /lib64/libssl.so.1.0.0 (0x00007fab627ff000)
>         libcrypto.so.1.0.0 => /lib64/libcrypto.so.1.0.0
> (0x00007fab6244e000) libdl.so.2 => /lib64/libdl.so.2 (0x00007fab6224a000)
>         libz.so.1 => /lib64/libz.so.1 (0x00007fab62032000)
>         libncurses.so.5 => /lib64/libncurses.so.5 (0x00007fab61ddd000)
>         libc.so.6 => /lib64/libc.so.6 (0x00007fab61a70000)
>         libstdc++.so.6 => /usr/lib64/libstdc++.so.6 (0x00007fab61767000)
>         libgcc_s.so.1 => /lib64/libgcc_s.so.1 (0x00007fab61551000)
>         libodbc.so.1 => /usr/lib64/libodbc.so.1 (0x00007fab612e5000)
>         /lib64/ld-linux-x86-64.so.2 (0x00007fab6390f000)


From justlikeef at gmail.com  Thu Jun 16 06:10:40 2011
From: justlikeef at gmail.com (Rob Hutton)
Date: Wed, 15 Jun 2011 22:10:40 -0400
Subject: [Freeswitch-users] SRTP
Message-ID: <201106152210.41334.justlikeef@gmail.com>

I think I have TLS and SRTP working at this point, but in the docs it says to use the following template for the dialplan:

http://wiki.freeswitch.org/wiki/Secure_RTP:

  <extension name="incoming-fxs">
    <condition field="destination_number" expression="^(202)$"/>
      <condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$" break="never">
      <action application="set" data="sip_secure_media=true"/>
      <action application="bridge" data="openzap/1/1"/>
    </condition>
  </extension

1) There is a missing > at the end of the close extension tag.
2) There is either a missing / at the end of the internal condition line, or a missing condition close tag somewhere
3) When I fix the interal condition, I get an error:

{ERR} mod_dialplan_xml.c:110 Nexted conditions are not allowed

All this, but a packet capture shows that SRTP is working based on what I did on:

http://wiki.freeswitch.org/wiki/SIP_TLS

Can someone give me some guidance on  the Secure_RTP page and I will update whatever?


From william at xofap.com  Thu Jun 16 07:14:20 2011
From: william at xofap.com (William Alianto)
Date: Thu, 16 Jun 2011 10:14:20 +0700
Subject: [Freeswitch-users] [Fwd:  Asterisk as gateway for FS]
In-Reply-To: <1308141679.16213.159.camel@luna.madrid.commsmundi.com>
References: <1308134265.2150.1.camel@DELL>
	<1308141679.16213.159.camel@luna.madrid.commsmundi.com>
Message-ID: <4DF9750C.5080107@xofap.com>

Do I need to do some change in the dialplan on both system?

On 2:59, Fran?ois Delawarde wrote:
> It is possible and works very well, just connect some SIP trunk between
> the two.
>
> Here we use it for TDM, IAX, and chan_mobile call termination.
>
> Regards,
> Fran?ois.
>
> On Wed, 2011-06-15 at 12:37 +0200, Henk Oegema wrote:
>> Just to be curios: What is the purpose of Asterisk being a gateway for
>> FS ?
>>
>> Rgds.
>> Henk
>>
>> -------- Forwarded Message --------
>> From: William Alianto<william at xofap.com>
>> Reply-to: FreeSWITCH Users Help
>> <freeswitch-users at lists.freeswitch.org>
>> To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
>> Subject: [Freeswitch-users] Asterisk as gateway for FS
>> Date: Wed, 15 Jun 2011 16:18:24 +0700
>>
>> Hi,
>>
>> I'm trying to setup a system using Asterisk and FreeSwitch, where
>> Asterisk will perform as gateway for FreeSwitch. Is this setup possible?
>> If yes, where can I find an example setup to look on to?
>>
>> -- 
>> Regards,
>>
>> William
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>


-- 
Regards,

William



From sid.kshatriya at gmail.com  Thu Jun 16 08:53:14 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Thu, 16 Jun 2011 10:23:14 +0530
Subject: [Freeswitch-users] Want to play music on hold while executing a
 time consuming operation in Lua
Message-ID: <BANLkTin5-d2QSCGCaaOW6j+Y_xu3AfNfog@mail.gmail.com>

I am implementing an IVR using Lua in Freeswitch. In my Lua script I use
curl to a web service. Sometimes the response takes a long time to come
back. During that time period I would like to play music on hold.

I have searched the freeswitch discussion archives a lot. There seem to be
many suggested ways to implement music on hold from a Lua script but the
answers are not very clear / not really applicable to my use case. I don't
know what method I should use and which one is recommended.

*Method 1:* Transfer the call to 9664 (music on hold extension). However the
implementation for this solution for this does not seem to be available in
Lua. For example: How would I transfer the call back?
*Method 2:* Using bgapi uuid_park park the call and using uuid_broadcast
play an audio file. Again what do I do to unpark the call..?

Thanks,

Sidharth
[P.S. This is a repost, I apologize but I never got any responses on my
previous email. Need help! :-) ]

-- 
Sidharth Kshatriya
www.sidk.info
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From sid.kshatriya at gmail.com  Thu Jun 16 09:03:24 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Thu, 16 Jun 2011 10:33:24 +0530
Subject: [Freeswitch-users] SRTP
In-Reply-To: <201106152210.41334.justlikeef@gmail.com>
References: <201106152210.41334.justlikeef@gmail.com>
Message-ID: <BANLkTi=tqCt__GXQr+xgpaKEZ24hx0Hz0A@mail.gmail.com>

Hi Rob,

I have adding the missing ">"

The wiki is freely editable. You could have made a new login account and
signed in and corrected it yourself.

Regarding your points 2, 3 -- I think no more edits are required apart from
just point 1.

Thanks,

Sidharth

On Thu, Jun 16, 2011 at 7:40 AM, Rob Hutton <justlikeef at gmail.com> wrote:

> I think I have TLS and SRTP working at this point, but in the docs it says
> to use the following template for the dialplan:
>
> http://wiki.freeswitch.org/wiki/Secure_RTP:
>
>  <extension name="incoming-fxs">
>    <condition field="destination_number" expression="^(202)$"/>
>      <condition field="${sip_has_crypto}"
> expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$"
> break="never">
>      <action application="set" data="sip_secure_media=true"/>
>      <action application="bridge" data="openzap/1/1"/>
>    </condition>
>  </extension
>
> 1) There is a missing > at the end of the close extension tag.
> 2) There is either a missing / at the end of the internal condition line,
> or a missing condition close tag somewhere
> 3) When I fix the interal condition, I get an error:
>
> {ERR} mod_dialplan_xml.c:110 Nexted conditions are not allowed
>
> All this, but a packet capture shows that SRTP is working based on what I
> did on:
>
> http://wiki.freeswitch.org/wiki/SIP_TLS
>
> Can someone give me some guidance on  the Secure_RTP page and I will update
> whatever?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sidharth Kshatriya
www.sidk.info
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From sid.kshatriya at gmail.com  Thu Jun 16 09:41:06 2011
From: sid.kshatriya at gmail.com (sidharth_k)
Date: Wed, 15 Jun 2011 22:41:06 -0700 (PDT)
Subject: [Freeswitch-users] Music On Hold
In-Reply-To: <AANLkTikeUuj8XHz2d9w3niXSO=wpptcQ-KSpdCqBefg1@mail.gmail.com>
References: <AANLkTin_qHEUnoFUo1jkX1xyWyyRiYe7sTE1P4czayUh@mail.gmail.com>
	<AANLkTikeUuj8XHz2d9w3niXSO=wpptcQ-KSpdCqBefg1@mail.gmail.com>
Message-ID: <1308202866242-6481837.post@n2.nabble.com>

I too have this requirement.

Lets say I have a script abc.lua 

In it, I

session:transfer("9664", "XML", "default")
session:setAutoHangup(false) -- prevent the script from ending

lets say I even have the uuid saved in a local variable uuid_value

After I do this, how do I bring the call back to the "context" of the script
abc.lua. In other words, how do I "untransfer" 


mercutioviz wrote:
> 
> Just x-fer the call to dp ext 9664. When you are done with the Lua tasks
> use
> uuid_transfer (or uuid_bridge) to grab the call out of 9664 and send it
> wherever you need it to go.
> 
> -MC
> 
> On Fri, Oct 22, 2010 at 12:19 AM, Chia-Yen Wu &lt;teddywu72 at gmail.com&gt;
> wrote:
> 
>> Hello,
>>
>> I am using freeswitch as an IVR server.
>> I would like to be able to put the call on hold and play music while
>> performing some other tasks in the lua.
>> when the task is complete , the music stop and the IVR continue
>>
>> but i cant find a way to do that, is there any function can do "music on
>> hold" in dialplan ?
>>
>> Thank you
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Music-On-Hold-tp5662660p6481837.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Thu Jun 16 11:15:33 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 16 Jun 2011 08:15:33 +0100
Subject: [Freeswitch-users] SRTP
In-Reply-To: <201106152210.41334.justlikeef@gmail.com>
References: <201106152210.41334.justlikeef@gmail.com>
Message-ID: <BANLkTi==5-N+HgmmU8kHaRq3SpfMAdP6cQ@mail.gmail.com>

 {ERR} mod_dialplan_xml.c:110 Nexted conditions are not allowed

That's because it shouldn't be nested. It's not missing a /, and the 1st
Should have the /. The extra indendation shouldn't be there on the 2nd.

It should look like this:

<extension name="incoming-fxs">
     <condition field="destination_number" expression="^(202)$"/>
     <condition field="${sip_has_crypto}"
expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$"
break="never">
          <action application="set" data="sip_secure_media=true"/>
          <action application="bridge" data="openzap/1/1"/>
     </condition>
</extension>

The two conditions function as an AND, even though it's not nested. FS stops
checking the extension as soon as it sees a condition that's false (at least
by default and in the above case), so if the destination is not 202 it'll
never get to the 2nd condition.

-Steve



On 16 June 2011 03:10, Rob Hutton <justlikeef at gmail.com> wrote:

> I think I have TLS and SRTP working at this point, but in the docs it says
> to use the following template for the dialplan:
>
> http://wiki.freeswitch.org/wiki/Secure_RTP:
>
>  <extension name="incoming-fxs">
>    <condition field="destination_number" expression="^(202)$"/>
>      <condition field="${sip_has_crypto}"
> expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$"
> break="never">
>      <action application="set" data="sip_secure_media=true"/>
>      <action application="bridge" data="openzap/1/1"/>
>    </condition>
>  </extension
>
> 1) There is a missing > at the end of the close extension tag.
> 2) There is either a missing / at the end of the internal condition line,
> or a missing condition close tag somewhere
> 3) When I fix the interal condition, I get an error:
>
> {ERR} mod_dialplan_xml.c:110 Nexted conditions are not allowed
>
> All this, but a packet capture shows that SRTP is working based on what I
> did on:
>
> http://wiki.freeswitch.org/wiki/SIP_TLS
>
> Can someone give me some guidance on  the Secure_RTP page and I will update
> whatever?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From helmut.kuper at ewetel.de  Thu Jun 16 11:40:47 2011
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Thu, 16 Jun 2011 09:40:47 +0200
Subject: [Freeswitch-users] Cleaning libs subdir for recompiling
Message-ID: <4DF9B37F.8080709@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello FS folks,

yesterday I copied a compiled FS git directory from 64Bit CentOS 5.5 to
a 64Bit CentOS 5.4 machine. This is, because I did some modifications in
serveral FS modules (e.g. openzap, freetdm, sofia, ...) and I was too
lazy to apply the modifications again to a fresh git clone, hehe.

I did a "make distclean" in the main directory. Then a bootstrap,
configure and make. I found that it seems that in the libs subdir were
still some information about the other sever's paths, which led to
compile errors. So I removed them as good as I could found the data.

My question is: Is there something like "make libsdistclean" or so which
resets the complete libs subdir of FS?


best regards
Helmut
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.10 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/

iEYEARECAAYFAk35s38ACgkQ4tZeNddg3dxeZACdF9+5UMmXwPBbIHzYhNjqaYen
1AMAn3JVy6UlLdEKFwUb9ljbaA83tIYU
=mIOI
-----END PGP SIGNATURE-----


From Stefan.Weigel at allianz-warranty.com  Thu Jun 16 12:01:40 2011
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Thu, 16 Jun 2011 10:01:40 +0200
Subject: [Freeswitch-users] unixODBC MySQL mysqld-5.0.77 - error
 messages with mod_callcenter
In-Reply-To: <BANLkTim3oqceLgziN75cVJAY5c78aoMBSA@mail.gmail.com>
References: <5003D7D3E06F514E8C682F18D223265C04D3B36C29@AZWSMS03.azwarranty.int>
	<BANLkTim3oqceLgziN75cVJAY5c78aoMBSA@mail.gmail.com>
Message-ID: <5003D7D3E06F514E8C682F18D223265C04D3B36C2A@AZWSMS03.azwarranty.int>

Hi,



sorry, I signed the message.

Again:



I recently changed from local DB to a ODBC connection with a MySQL backend. After testing around I'm now getting the following errors:



http://pastebin.freeswitch.org/16503





When executing the statements for example with phpMyAdmin I'm not getting any error. Any hints on this topic ?





Thanks and best regards



Stefan







I also found some related topics on the list, so I fixed this issue with setting the options parameter in odbc.ini to allow multiple statements.





Thanks and best regards



Stefan



Stefan Weigel
Advanced IT-Professional

Tel.: +49 89 2000 48 975
Fax: +49 89 2000 48 566
eMail: Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>

Allianz Automotive Services GmbH

Einsteinring 28
85609 Aschheim
Germany

http://www.allianz-warranty.com
Gesch?ftsf?hrung: Andreas R?sing, Horst Ziegler
Amtsgericht M?nchen, HRB 175682
F?r Umsatzsteuerzwecke: Ust-ID-Nr.: DE 262 617 720



Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Giovanni Maruzzelli
Gesendet: Mittwoch, 15. Juni 2011 22:22
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] unixODBC MySQL mysqld-5.0.77 - error messages with mod_callcenter



Stefan,
your mail is unreadable.
Send plain old pure ASCII, not fancy cryptostuff :)
-giovanni

On Wed, Jun 15, 2011 at 5:14 PM, Weigel, Stefan <Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>> wrote:

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




--
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

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From Stefan.Weigel at allianz-warranty.com  Thu Jun 16 12:30:08 2011
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Thu, 16 Jun 2011 10:30:08 +0200
Subject: [Freeswitch-users] mod_spandsp event spandsp::rxfaxresult variables
	missing
Message-ID: <5003D7D3E06F514E8C682F18D223265C04D3B36C2B@AZWSMS03.azwarranty.int>

Hi list,



yesterday I had a ODBC related problem. Therefore I used the latest snapshot. Now my scripts are not working properly (I'm doing an ESL connection and listening for spandsp::rxfaxresults events).

With an older release (mid of may) I was getting lot's more info's (for example variable_current_application_data), now I'm only getting the information documented on the wiki page:



fax-document-transferred-pages

fax-document-total-pages

fax-image-resolution

fax-image-size

fax-bad-rows

fax-transfer-rate

fax-result-code

fax-result-text

fax-ecm-used

fax-local-station-id

fax-remote-station-id





What changed and how can I get back the needed information ?







Thanks and best regards



Stefan

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From steveayre at gmail.com  Thu Jun 16 13:46:53 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 16 Jun 2011 10:46:53 +0100
Subject: [Freeswitch-users] mod_spandsp event spandsp::rxfaxresult
 variables missing
In-Reply-To: <5003D7D3E06F514E8C682F18D223265C04D3B36C2B@AZWSMS03.azwarranty.int>
References: <5003D7D3E06F514E8C682F18D223265C04D3B36C2B@AZWSMS03.azwarranty.int>
Message-ID: <BANLkTin3vQii2arQd4KhmSBDoJFivaLiBQ@mail.gmail.com>

Do you know the version number of the older working release?

http://jira.freeswitch.org/browse/FS-3004
Looks like Flavio's patch wasn't applied entirely on May 3rd - it's missing
the "switch_channel_event_set_data(channel, event);" line. Not sure if
that's what's causing it, but it seems likely. The event generation hasn't
been modified since then.

-Steve




On 16 June 2011 09:30, Weigel, Stefan <Stefan.Weigel at allianz-warranty.com>wrote:

>  Hi list,
>
>
>
> yesterday I had a ODBC related problem. Therefore I used the latest
> snapshot. Now my scripts are not working properly (I?m doing an ESL
> connection and listening for spandsp::rxfaxresults events).
>
> With an older release (mid of may) I was getting lot?s more info?s (for
> example variable_current_application_data), now I?m only getting the
> information documented on the wiki page:
>
>
>
> fax-document-transferred-pages
>
> fax-document-total-pages
>
> fax-image-resolution
>
> fax-image-size
>
> fax-bad-rows
>
> fax-transfer-rate
>
> fax-result-code
>
> fax-result-text
>
> fax-ecm-used
>
> fax-local-station-id
>
> fax-remote-station-id
>
>
>
>
>
> What changed and how can I get back the needed information ?
>
>
>
>
>
>
>
> Thanks and best regards
>
>
>
> Stefan
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Stefan.Weigel at allianz-warranty.com  Thu Jun 16 14:24:45 2011
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Thu, 16 Jun 2011 12:24:45 +0200
Subject: [Freeswitch-users] mod_spandsp event spandsp::rxfaxresult
 variables missing
In-Reply-To: <BANLkTin3vQii2arQd4KhmSBDoJFivaLiBQ@mail.gmail.com>
References: <5003D7D3E06F514E8C682F18D223265C04D3B36C2B@AZWSMS03.azwarranty.int>
	<BANLkTin3vQii2arQd4KhmSBDoJFivaLiBQ@mail.gmail.com>
Message-ID: <5003D7D3E06F514E8C682F18D223265C04D3B36C2E@AZWSMS03.azwarranty.int>

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From Stefan.Weigel at allianz-warranty.com  Thu Jun 16 14:25:18 2011
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Thu, 16 Jun 2011 12:25:18 +0200
Subject: [Freeswitch-users] mod_spandsp event spandsp::rxfaxresult
 variables missing
In-Reply-To: <BANLkTin3vQii2arQd4KhmSBDoJFivaLiBQ@mail.gmail.com>
References: <5003D7D3E06F514E8C682F18D223265C04D3B36C2B@AZWSMS03.azwarranty.int>
	<BANLkTin3vQii2arQd4KhmSBDoJFivaLiBQ@mail.gmail.com>
Message-ID: <5003D7D3E06F514E8C682F18D223265C04D3B36C2F@AZWSMS03.azwarranty.int>

Hi Steve,



version shows me only 'git-.

I'm pretty sure that it was the snapshot downloaded on the 2nd of May. But I can remember also applied the patches.

I checked and the line



                               switch_channel_event_set_data(channel, event);



is missing. I will rebuild the module and test again and tell you more.





Thanks and best regards



Stefan





Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Steven Ayre
Gesendet: Donnerstag, 16. Juni 2011 11:47
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] mod_spandsp event spandsp::rxfaxresult variables missing



Do you know the version number of the older working release?



http://jira.freeswitch.org/browse/FS-3004

Looks like Flavio's patch wasn't applied entirely on May 3rd - it's missing the "switch_channel_event_set_data(channel, event);" line. Not sure if that's what's causing it, but it seems likely. The event generation hasn't been modified since then.



-Steve









On 16 June 2011 09:30, Weigel, Stefan <Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>> wrote:

Hi list,



yesterday I had a ODBC related problem. Therefore I used the latest snapshot. Now my scripts are not working properly (I'm doing an ESL connection and listening for spandsp::rxfaxresults events).

With an older release (mid of may) I was getting lot's more info's (for example variable_current_application_data), now I'm only getting the information documented on the wiki page:



fax-document-transferred-pages

fax-document-total-pages

fax-image-resolution

fax-image-size

fax-bad-rows

fax-transfer-rate

fax-result-code

fax-result-text

fax-ecm-used

fax-local-station-id

fax-remote-station-id





What changed and how can I get back the needed information ?







Thanks and best regards



Stefan


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



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From david.ponzone at ipeva.fr  Thu Jun 16 14:41:00 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 16 Jun 2011 12:41:00 +0200
Subject: [Freeswitch-users] Want to play music on hold while executing a
	time consuming operation in Lua
In-Reply-To: <BANLkTin5-d2QSCGCaaOW6j+Y_xu3AfNfog@mail.gmail.com>
References: <BANLkTin5-d2QSCGCaaOW6j+Y_xu3AfNfog@mail.gmail.com>
Message-ID: <30FA9F9A-6489-4954-9DAA-39152DC021A4@ipeva.fr>

Sidharth,

Generally, if noone answers, it means noone has an interesting answer or noone had the time yet to answer.
It's quite like when you ask a girl out. If she doesn't answer, it's unlikely harassing her will improve the situation ... most of the times :)

You can also try to go on irc #freeswitch.
And also, you can read the wiki.

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

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

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




Le 16/06/2011 ? 06:53, Sidharth Kshatriya a ?crit :

> I am implementing an IVR using Lua in Freeswitch. In my Lua script I use curl to a web service. Sometimes the response takes a long time to come back. During that time period I would like to play music on hold.
> 
> I have searched the freeswitch discussion archives a lot. There seem to be many suggested ways to implement music on hold from a Lua script but the answers are not very clear / not really applicable to my use case. I don't know what method I should use and which one is recommended.
> 
> Method 1: Transfer the call to 9664 (music on hold extension). However the implementation for this solution for this does not seem to be available in Lua. For example: How would I transfer the call back?
> Method 2: Using bgapi uuid_park park the call and using uuid_broadcast play an audio file. Again what do I do to unpark the call..?
> 
> Thanks,
> 
> Sidharth
> [P.S. This is a repost, I apologize but I never got any responses on my previous email. Need help! :-) ]
>  
> -- 
> Sidharth Kshatriya
> www.sidk.info
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From sascha.daniels at amooma.de  Thu Jun 16 14:56:44 2011
From: sascha.daniels at amooma.de (Sascha Daniels)
Date: Thu, 16 Jun 2011 12:56:44 +0200
Subject: [Freeswitch-users] mod_fifo shared lag
Message-ID: <4DF9E16C.4090608@amooma.de>

Hi together.

Is there a possibility to have a shared lag over several fifos?

In a callcenter with agents in several Queues the lag is 10sec. After a
call from fifo A the agent should not get a call from fifo B within
those 10sec.

Basically like shared_lastcall=yes in Asterisk 1.6

Regards

                    Sascha

-- 
AMOOMA GmbH - Bachstr. 124 - 56566 Neuwied  -->  http://www.amooma.de
Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister Montabaur B14998

B?cher: http://das-asterisk-buch.de - http://ruby-auf-schienen.de



From steveayre at gmail.com  Thu Jun 16 15:01:56 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 16 Jun 2011 12:01:56 +0100
Subject: [Freeswitch-users] mod_spandsp event spandsp::rxfaxresult
 variables missing
In-Reply-To: <5003D7D3E06F514E8C682F18D223265C04D3B36C2F@AZWSMS03.azwarranty.int>
References: <5003D7D3E06F514E8C682F18D223265C04D3B36C2B@AZWSMS03.azwarranty.int>
	<BANLkTin3vQii2arQd4KhmSBDoJFivaLiBQ@mail.gmail.com>
	<5003D7D3E06F514E8C682F18D223265C04D3B36C2F@AZWSMS03.azwarranty.int>
Message-ID: <BANLkTi=-+Ob_F065O-mY8iS819QPJJxw+w@mail.gmail.com>

Ok. If they're still missing after you rebuild you could reopen the Jira.

-Steve


On 16 June 2011 11:25, Weigel, Stefan <Stefan.Weigel at allianz-warranty.com>wrote:

>  Hi Steve,
>
>
>
> version shows me only ?git-.
>
> I?m pretty sure that it was the snapshot downloaded on the 2nd of May. But
> I can remember also applied the patches.
>
> I checked and the line
>
>
>
>                                switch_channel_event_set_data(channel,
> event);
>
>
>
> is missing. I will rebuild the module and test again and tell you more.
>
>
>
>
>
> Thanks and best regards
>
>
>
> Stefan
>
>
>
>
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Steven
> Ayre
> *Gesendet:* Donnerstag, 16. Juni 2011 11:47
> *An:* FreeSWITCH Users Help
> *Betreff:* Re: [Freeswitch-users] mod_spandsp event spandsp::rxfaxresult
> variables missing
>
>
>
> Do you know the version number of the older working release?
>
>
>
> http://jira.freeswitch.org/browse/FS-3004
>
> Looks like Flavio's patch wasn't applied entirely on May 3rd - it's missing
> the "switch_channel_event_set_data(channel, event);" line. Not sure if
> that's what's causing it, but it seems likely. The event generation hasn't
> been modified since then.
>
>
>
> -Steve
>
>
>
>
>
>
>
>
>
> On 16 June 2011 09:30, Weigel, Stefan <Stefan.Weigel at allianz-warranty.com>
> wrote:
>
> Hi list,
>
>
>
> yesterday I had a ODBC related problem. Therefore I used the latest
> snapshot. Now my scripts are not working properly (I?m doing an ESL
> connection and listening for spandsp::rxfaxresults events).
>
> With an older release (mid of may) I was getting lot?s more info?s (for
> example variable_current_application_data), now I?m only getting the
> information documented on the wiki page:
>
>
>
> fax-document-transferred-pages
>
> fax-document-total-pages
>
> fax-image-resolution
>
> fax-image-size
>
> fax-bad-rows
>
> fax-transfer-rate
>
> fax-result-code
>
> fax-result-text
>
> fax-ecm-used
>
> fax-local-station-id
>
> fax-remote-station-id
>
>
>
>
>
> What changed and how can I get back the needed information ?
>
>
>
>
>
>
>
> Thanks and best regards
>
>
>
> Stefan
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sharad at coraltele.com  Thu Jun 16 15:54:54 2011
From: sharad at coraltele.com (sharad)
Date: Thu, 16 Jun 2011 17:24:54 +0530
Subject: [Freeswitch-users] Originate API - Context remain default till the
	call is answered
References: <mailman.10614.1308221492.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <57087ED9B5CB4F35AC110A4415C4A6B9@sharad>

Hi All

When we make an API originate with desired context say `abcd', `show 
channels' shows the context `default' before answer. But as soon as call is 
answered, context also gets changed to `abcd'.

Is there any way to make the context `abcd' before call answer also.

Thanks
Regards
Sharad 



From kris at kriskinc.com  Thu Jun 16 17:32:16 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 16 Jun 2011 09:32:16 -0400
Subject: [Freeswitch-users] 183 after 180
In-Reply-To: <EC6AC96E-7FAC-4BB4-9FF9-1711CB33094C@teliax.com>
References: <9AD5D885-6663-4C1D-B3FA-7CFC1C49F68F@teliax.com>
	<BANLkTin+fvcVSfiaAXb+vCETTV387UL+XA@mail.gmail.com>
	<EC6AC96E-7FAC-4BB4-9FF9-1711CB33094C@teliax.com>
Message-ID: <BANLkTi=ywUGzi-TLhW=0NB-egK8z=c2BEQ@mail.gmail.com>

David,

  I was going to suggest bypass_media_after_bridge (re-INVITE) but
I've never used it so I wasn't sure how well it worked.  This is still
a very interesting situation...

  One question: How do you detect the remote end has responded with a
180/183 w/o SDP in order to execute these commands conditionally?

On Wed, Jun 15, 2011 at 6:35 PM, David Aldworth <daldworth at teliax.com> wrote:
> Your comments about different endpoints reacting different is my concern exactly. That said, we came up with a fairly simple solution (below) that, thus far, seems alright. However, not all scenarios have been fully test, such as what happens if we get an early media operator intercept with the 183, what happens if the endpoint rejects the re-invite, etc.
>
> <action application="set" data="bypass_media=false"/>
> <action application="set" data="ringback=$${us-ring}"/>
> <action application="set" data="bypass_media_after_bridge=true"/>

-- 
Kristian Kielhofner


From kris at kriskinc.com  Thu Jun 16 17:42:30 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 16 Jun 2011 09:42:30 -0400
Subject: [Freeswitch-users] Funky Gateway Confusion?
In-Reply-To: <BANLkTi=6909x6oK_dYEm+u2tZy4oeyP47A@mail.gmail.com>
References: <BANLkTi=6909x6oK_dYEm+u2tZy4oeyP47A@mail.gmail.com>
Message-ID: <BANLkTikrj4_j1nTkoEF1o5807qP+Na6Rrw@mail.gmail.com>

Avi,

  Based on the SDP from the 200 OK I'd say your provider is using a
proxy in the SER family.  The SER proxies allow the proxy admin to
configure the nathelper "pinger" address:

http://www.opensips.org/html/docs/modules/1.6.x/nathelper.html#id292784

  They probably just couldn't think of anything better than "1111".
As far as "Spikko001", I have no idea...

On Wed, Jun 15, 2011 at 6:49 PM, Avi Marcus <avi at avimarcus.net> wrote:
> When making a call out on a gateway named "xconnect" they ping to see if my
> machine is still active.
> It uses 1111 in there, which happens to be the highest numbered SIP username
> on my system.
> Oh, and?gw=Spikko001?gets in there too which is a completely unrelated
> gateway on my system.
> What in the world is going on?
> http://pastebin.freeswitch.org/16519
> It doesn't seem to be causing any problems.. but it's so weird.
> kinda old: FreeSWITCH Version 1.0.head (git-0911ed7 2011-03-05 12-49-19
> -0500)
>
> -Avi Marcus
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner


From kris at kriskinc.com  Thu Jun 16 17:51:59 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 16 Jun 2011 09:51:59 -0400
Subject: [Freeswitch-users] in bypass_media mode,
 2833 payload type modified by freeswitch!
In-Reply-To: <BANLkTimLLbr+oEh3m0JY7YRSWMDQzP5ABQ@mail.gmail.com>
References: <BANLkTi=HpPE7q4Kbrkwe-KJxj69SYJtx0Q@mail.gmail.com>
	<4A5154F6-90DE-4B40-9F24-E3E0D0824D02@ipeva.fr>
	<BANLkTikfoYbL-atXM78W7BTLe3LNUheF9A@mail.gmail.com>
	<BANLkTikMc=Pdwx6W85JBHZ4AJxpKhdhz5A@mail.gmail.com>
	<BANLkTikSr75c8cucBZcL+bkkTisuy6X1zw@mail.gmail.com>
	<BANLkTi=F2U+VPJGdRWjot6wTktZrF3ZCAg@mail.gmail.com>
	<BANLkTikhE7Mj0nM4J0+NN+zEOfVjaoiOCQ@mail.gmail.com>
	<BANLkTimLLbr+oEh3m0JY7YRSWMDQzP5ABQ@mail.gmail.com>
Message-ID: <BANLkTinwxsVVLVGqzey-ygdkwFpfZtjBOQ@mail.gmail.com>

I didn't know about this parameter either.  Is it new?

I'd argue that it should be enabled by default when bypass_media is in
enabled.  If not, it leaves open cases such as this one where you
can't win - some parameters of the SDP will be changed but not others?

If FreeSWITCH is completely rewriting the SDP and handling the media
to account for differences in the RTP streams that's fine (normal
media mode).

If FreeSWITCH is rewriting some SDP params without handling the media
it leaves the endpoints in a broken state that no one would ever want.
 In this case FreeSWITCH changed the SDP to say 2833 is payload type
101 when the RTP stream from the remote end would clearly have it at
payload type 110!  The endpoints would end up completely confused...

I think this parameter has some purpose and I always enjoy having
extra knobs to turn.  However, it should be enabled by default when
bypass_media is in use (to be disabled selectively by users in cases
where they need to, for some reason).

What about proxy_media?  As far as I can tell the same thing would
happen there (or worse).  The SDP would (probably) get changed and the
media would be proxied through FreeSWITCH but the RTP and SDP would
not agree on payload type (same as now).  From what it sounds like
this soa parameter is not compatible with proxy_media (hands off SDP)
so what would happen there?

On Wed, Jun 15, 2011 at 7:29 PM, Michael Collins <msc at freeswitch.org> wrote:
> And FTR, this *is* on the wiki - I just didn't know about it. :)
> http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation
> (Last item)
> -MC

-- 
Kristian Kielhofner


From mehmasarja at gmail.com  Thu Jun 16 17:57:36 2011
From: mehmasarja at gmail.com (Mehma Sarja)
Date: Thu, 16 Jun 2011 06:57:36 -0700
Subject: [Freeswitch-users] Want to play music on hold while executing a
 time consuming operation in Lua
In-Reply-To: <30FA9F9A-6489-4954-9DAA-39152DC021A4@ipeva.fr>
References: <BANLkTin5-d2QSCGCaaOW6j+Y_xu3AfNfog@mail.gmail.com>
	<30FA9F9A-6489-4954-9DAA-39152DC021A4@ipeva.fr>
Message-ID: <4DFA0BD0.7010104@gmail.com>

This is true. Although I've never formally asked a girl out, I have 
posted a lame question and waited for days until someone responded. Then 
we have a few messages back and forth. Maybe Sidharth has a point 
because now I'm married and have lost the interest to ask anyone out, if 
I had it in the first place.

Mehma
===
On 6/16/11 3:41 AM, David Ponzone wrote:
>
> Generally, if noone answers, it means noone has an interesting answer 
> or noone had the time yet to answer.
> It's quite like when you ask a girl out. If she doesn't answer, it's 
> unlikely harassing her will improve the situation ... most of the times :)
>
>
>
> Le 16/06/2011 ? 06:53, Sidharth Kshatriya a ?crit :
>
>> [P.S. This is a repost, I apologize but I never got any responses on 
>> my previous email. Need help! :-) ]

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From anthony.minessale at gmail.com  Thu Jun 16 19:10:32 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 16 Jun 2011 10:10:32 -0500
Subject: [Freeswitch-users] Presence SQL Error
In-Reply-To: <BANLkTi=Gk4DT6skr2Kz2QwEjPmvRSy7vPw@mail.gmail.com>
References: <BANLkTi=Gk4DT6skr2Kz2QwEjPmvRSy7vPw@mail.gmail.com>
Message-ID: <BANLkTikcCeoq_MV-SOLWGkxGjzx96vSg2w@mail.gmail.com>

this is the result of debug-presence enabled on your profile.
It puts all the sql etc into error level so it stands out.


On Wed, Jun 15, 2011 at 5:44 PM, Avi Marcus <avi at avimarcus.net> wrote:
> This is a pretty fresh install of FS on centos, using sqlite for the DB.
> I do have multiple domains set up.
> There's an SQL error for the presence requests, but FS isn't telling me much
> info. What can I do about this?
> The only times I've used presence, it basically just worked.
> http://pastebin.freeswitch.org/16518
> -Avi
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From jeff at jefflenk.com  Thu Jun 16 20:17:00 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 16 Jun 2011 09:17:00 -0700 (PDT)
Subject: [Freeswitch-users] mod_spandsp event spandsp::rxfaxresult
	variables missing
In-Reply-To: <BANLkTi=-+Ob_F065O-mY8iS819QPJJxw+w@mail.gmail.com>
References: <5003D7D3E06F514E8C682F18D223265C04D3B36C2B@AZWSMS03.azwarranty.int>
	<BANLkTin3vQii2arQd4KhmSBDoJFivaLiBQ@mail.gmail.com>
	<5003D7D3E06F514E8C682F18D223265C04D3B36C2F@AZWSMS03.azwarranty.int>
	<BANLkTi=-+Ob_F065O-mY8iS819QPJJxw+w@mail.gmail.com>
Message-ID: <1308241020809-6483715.post@n2.nabble.com>

Check git head for recent changes to the fax event headers

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-spandsp-event-spandsp-rxfaxresult-variables-missing-tp6482195p6483715.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From ankitwalia4u at gmail.com  Thu Jun 16 21:04:12 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Thu, 16 Jun 2011 22:34:12 +0530
Subject: [Freeswitch-users] Making a external call to Mobile number
Message-ID: <BANLkTinFuhpjGkLv14UC3LGUYcFc8H9cjQ@mail.gmail.com>

Dear all,

I am trying to make an external call to a mobile number of India using the
default ext number 1005. For doing this, I did the following things.

First, I created a user at IPTEL.org, I added a SIP profile. I registered
with IPTEL. I also added a dial plan regex for 91 - (10 digit number). But,
because of some issue, I was not able to connect to IPTEL SIP. I was getting
error on X-LITE about some server issue.

then, I tried my VOIP account with ActionVoip which is a paid account and
has SIP service. I could connect to my action voip account through X-LITE
SIP.

I made a SIP profile for Action Voip and registered the username with action
voip in FS. On reload, it got registered using action voip SIP profile. I
changed the dialplan to use actionvoip gateway.

But still, when I am trying to call to my mobile number in India. The call
is not able to connect.

I think there is gap in my knowledge. Please help me I am new to FreeSwitch.

Thanks
Ankit

-- 
Life is like a rose its upto u feel it as its fragrance or thorns
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From justlikeef at gmail.com  Thu Jun 16 21:22:53 2011
From: justlikeef at gmail.com (Rob Hutton)
Date: Thu, 16 Jun 2011 13:22:53 -0400
Subject: [Freeswitch-users] SRTP
In-Reply-To: <BANLkTi==5-N+HgmmU8kHaRq3SpfMAdP6cQ@mail.gmail.com>
References: <201106152210.41334.justlikeef@gmail.com>
	<BANLkTi==5-N+HgmmU8kHaRq3SpfMAdP6cQ@mail.gmail.com>
Message-ID: <201106161322.53638.justlikeef@gmail.com>

Steven -

Thanks for the help here...

So there would have to be two dialplan entries for this number to work with 
either RTP or SRTP? (Maybe two devices registering to the same user?)

Would it make more since to do this in a more global manner higher up in the 
dialplan in its own condition block?


On Thursday 16 June 2011 03:15:33 Steven Ayre wrote:
>  {ERR} mod_dialplan_xml.c:110 Nexted conditions are not allowed
> 
> That's because it shouldn't be nested. It's not missing a /, and the 1st
> Should have the /. The extra indendation shouldn't be there on the 2nd.
> 
> It should look like this:
> 
> <extension name="incoming-fxs">
>      <condition field="destination_number" expression="^(202)$"/>
>      <condition field="${sip_has_crypto}"
> expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$"
> break="never">
>           <action application="set" data="sip_secure_media=true"/>
>           <action application="bridge" data="openzap/1/1"/>
>      </condition>
> </extension>
> 
> The two conditions function as an AND, even though it's not nested. FS
> stops checking the extension as soon as it sees a condition that's false
> (at least by default and in the above case), so if the destination is not
> 202 it'll never get to the 2nd condition.
> 
> -Steve
> 
> On 16 June 2011 03:10, Rob Hutton <justlikeef at gmail.com> wrote:
> > I think I have TLS and SRTP working at this point, but in the docs it
> > says to use the following template for the dialplan:
> > 
> > http://wiki.freeswitch.org/wiki/Secure_RTP:
> >  <extension name="incoming-fxs">
> >  
> >    <condition field="destination_number" expression="^(202)$"/>
> >    
> >      <condition field="${sip_has_crypto}"
> > 
> > expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$"
> > break="never">
> > 
> >      <action application="set" data="sip_secure_media=true"/>
> >      <action application="bridge" data="openzap/1/1"/>
> >    
> >    </condition>
> >  
> >  </extension
> > 
> > 1) There is a missing > at the end of the close extension tag.
> > 2) There is either a missing / at the end of the internal condition line,
> > or a missing condition close tag somewhere
> > 3) When I fix the interal condition, I get an error:
> > 
> > {ERR} mod_dialplan_xml.c:110 Nexted conditions are not allowed
> > 
> > All this, but a packet capture shows that SRTP is working based on what I
> > did on:
> > 
> > http://wiki.freeswitch.org/wiki/SIP_TLS
> > 
> > Can someone give me some guidance on  the Secure_RTP page and I will
> > update whatever?
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org


From brian at freeswitch.org  Thu Jun 16 21:23:42 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 16 Jun 2011 12:23:42 -0500
Subject: [Freeswitch-users] 183 after 180
In-Reply-To: <BANLkTi=ywUGzi-TLhW=0NB-egK8z=c2BEQ@mail.gmail.com>
References: <9AD5D885-6663-4C1D-B3FA-7CFC1C49F68F@teliax.com>
	<BANLkTin+fvcVSfiaAXb+vCETTV387UL+XA@mail.gmail.com>
	<EC6AC96E-7FAC-4BB4-9FF9-1711CB33094C@teliax.com>
	<BANLkTi=ywUGzi-TLhW=0NB-egK8z=c2BEQ@mail.gmail.com>
Message-ID: <38D35468-79AF-42AA-9A1D-DBFAD3600200@freeswitch.org>

You could use execute_on_media to set a variable ;)

/b

On Jun 16, 2011, at 8:32 AM, Kristian Kielhofner wrote:

>  One question: How do you detect the remote end has responded with a
> 180/183 w/o SDP in order to execute these commands conditionally?



From kris at kriskinc.com  Thu Jun 16 21:28:04 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 16 Jun 2011 13:28:04 -0400
Subject: [Freeswitch-users] 183 after 180
In-Reply-To: <38D35468-79AF-42AA-9A1D-DBFAD3600200@freeswitch.org>
References: <9AD5D885-6663-4C1D-B3FA-7CFC1C49F68F@teliax.com>
	<BANLkTin+fvcVSfiaAXb+vCETTV387UL+XA@mail.gmail.com>
	<EC6AC96E-7FAC-4BB4-9FF9-1711CB33094C@teliax.com>
	<BANLkTi=ywUGzi-TLhW=0NB-egK8z=c2BEQ@mail.gmail.com>
	<38D35468-79AF-42AA-9A1D-DBFAD3600200@freeswitch.org>
Message-ID: <BANLkTimikXTw3H6ng2La1kr4WPbO4kiN=A@mail.gmail.com>

According to the wiki both execute_on_media and execute_on_preanswer
don't distinguish between the various 180/183 with or without SDP
scenarios.

I guess I just may have to try it :).

On Thu, Jun 16, 2011 at 1:23 PM, Brian West <brian at freeswitch.org> wrote:
> You could use execute_on_media to set a variable ;)
>
> /b

-- 
Kristian Kielhofner


From anthony.minessale at gmail.com  Thu Jun 16 21:37:05 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 16 Jun 2011 12:37:05 -0500
Subject: [Freeswitch-users] 183 after 180
In-Reply-To: <BANLkTimikXTw3H6ng2La1kr4WPbO4kiN=A@mail.gmail.com>
References: <9AD5D885-6663-4C1D-B3FA-7CFC1C49F68F@teliax.com>
	<BANLkTin+fvcVSfiaAXb+vCETTV387UL+XA@mail.gmail.com>
	<EC6AC96E-7FAC-4BB4-9FF9-1711CB33094C@teliax.com>
	<BANLkTi=ywUGzi-TLhW=0NB-egK8z=c2BEQ@mail.gmail.com>
	<38D35468-79AF-42AA-9A1D-DBFAD3600200@freeswitch.org>
	<BANLkTimikXTw3H6ng2La1kr4WPbO4kiN=A@mail.gmail.com>
Message-ID: <BANLkTimSSP-OTeVk7=HoJjFYWiAKaVrXeA@mail.gmail.com>

if you set ringback variable on A and bypass_media_after_bridge=true
and don't set ignore_early_media
you should get what you want.
The generate will only happen on 180 and the 183 would bridge.

and in both cases once it's answered it goes to no media mode.


On Thu, Jun 16, 2011 at 12:28 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
> According to the wiki both execute_on_media and execute_on_preanswer
> don't distinguish between the various 180/183 with or without SDP
> scenarios.
>
> I guess I just may have to try it :).
>
> On Thu, Jun 16, 2011 at 1:23 PM, Brian West <brian at freeswitch.org> wrote:
>> You could use execute_on_media to set a variable ;)
>>
>> /b
>
> --
> Kristian Kielhofner
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From daldworth at teliax.com  Thu Jun 16 21:42:32 2011
From: daldworth at teliax.com (David Aldworth)
Date: Thu, 16 Jun 2011 11:42:32 -0600
Subject: [Freeswitch-users] 183 after 180
In-Reply-To: <BANLkTimSSP-OTeVk7=HoJjFYWiAKaVrXeA@mail.gmail.com>
References: <9AD5D885-6663-4C1D-B3FA-7CFC1C49F68F@teliax.com>
	<BANLkTin+fvcVSfiaAXb+vCETTV387UL+XA@mail.gmail.com>
	<EC6AC96E-7FAC-4BB4-9FF9-1711CB33094C@teliax.com>
	<BANLkTi=ywUGzi-TLhW=0NB-egK8z=c2BEQ@mail.gmail.com>
	<38D35468-79AF-42AA-9A1D-DBFAD3600200@freeswitch.org>
	<BANLkTimikXTw3H6ng2La1kr4WPbO4kiN=A@mail.gmail.com>
	<BANLkTimSSP-OTeVk7=HoJjFYWiAKaVrXeA@mail.gmail.com>
Message-ID: <930F2905-EDD4-43C5-AB0D-BE132977FE0E@teliax.com>

Thank you. That is indeed the scenario I was trying to confirm. So far I've had no problem with the dp settings of anchoring, then setting US ring, then bypassing after bridge. Thanks to Math for help in the forum too! He suggested I add the US ring and that without it FS would just relay the 180.

The execute_on_media/preanswer has interesting implications too...

David

On Jun 16, 2011, at 11:37 AM, Anthony Minessale wrote:

> if you set ringback variable on A and bypass_media_after_bridge=true
> and don't set ignore_early_media
> you should get what you want.
> The generate will only happen on 180 and the 183 would bridge.
> 
> and in both cases once it's answered it goes to no media mode.
> 
> 
> On Thu, Jun 16, 2011 at 12:28 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
>> According to the wiki both execute_on_media and execute_on_preanswer
>> don't distinguish between the various 180/183 with or without SDP
>> scenarios.
>> 
>> I guess I just may have to try it :).
>> 
>> On Thu, Jun 16, 2011 at 1:23 PM, Brian West <brian at freeswitch.org> wrote:
>>> You could use execute_on_media to set a variable ;)
>>> 
>>> /b
>> 
>> --
>> Kristian Kielhofner
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Thu Jun 16 21:44:41 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 16 Jun 2011 12:44:41 -0500
Subject: [Freeswitch-users] 183 after 180
In-Reply-To: <BANLkTimikXTw3H6ng2La1kr4WPbO4kiN=A@mail.gmail.com>
References: <9AD5D885-6663-4C1D-B3FA-7CFC1C49F68F@teliax.com>
	<BANLkTin+fvcVSfiaAXb+vCETTV387UL+XA@mail.gmail.com>
	<EC6AC96E-7FAC-4BB4-9FF9-1711CB33094C@teliax.com>
	<BANLkTi=ywUGzi-TLhW=0NB-egK8z=c2BEQ@mail.gmail.com>
	<38D35468-79AF-42AA-9A1D-DBFAD3600200@freeswitch.org>
	<BANLkTimikXTw3H6ng2La1kr4WPbO4kiN=A@mail.gmail.com>
Message-ID: <A02BECF5-BDCC-4F59-A216-D18B1CB5499D@freeswitch.org>

Hrm.. that is true you can get a 180 with an SDP which we treat like a 183 but hrm I think you're correct.

/b

On Jun 16, 2011, at 12:28 PM, Kristian Kielhofner wrote:

> According to the wiki both execute_on_media and execute_on_preanswer
> don't distinguish between the various 180/183 with or without SDP
> scenarios.
> 
> I guess I just may have to try it :).
> 
> On Thu, Jun 16, 2011 at 1:23 PM, Brian West <brian at freeswitch.org> wrote:
>> You could use execute_on_media to set a variable ;)
>> 
>> /b
> 

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From qingquan at globalroam.com  Thu Jun 16 05:46:37 2011
From: qingquan at globalroam.com (qingquan luo)
Date: Thu, 16 Jun 2011 09:46:37 +0800
Subject: [Freeswitch-users] in bypass_media mode,
 2833 payload type modified by freeswitch!
In-Reply-To: <BANLkTimLLbr+oEh3m0JY7YRSWMDQzP5ABQ@mail.gmail.com>
References: <BANLkTi=HpPE7q4Kbrkwe-KJxj69SYJtx0Q@mail.gmail.com>
	<4A5154F6-90DE-4B40-9F24-E3E0D0824D02@ipeva.fr>
	<BANLkTikfoYbL-atXM78W7BTLe3LNUheF9A@mail.gmail.com>
	<BANLkTikMc=Pdwx6W85JBHZ4AJxpKhdhz5A@mail.gmail.com>
	<BANLkTikSr75c8cucBZcL+bkkTisuy6X1zw@mail.gmail.com>
	<BANLkTi=F2U+VPJGdRWjot6wTktZrF3ZCAg@mail.gmail.com>
	<BANLkTikhE7Mj0nM4J0+NN+zEOfVjaoiOCQ@mail.gmail.com>
	<BANLkTimLLbr+oEh3m0JY7YRSWMDQzP5ABQ@mail.gmail.com>
Message-ID: <BANLkTin=NAM9CBkUH8dNs72Cgoji6VTutw@mail.gmail.com>

Hi All,

    Thank you all for spend time on this question.
    I has try use export sip_enable_soa=false, And It work perfectly.

Thanks a lot.

Respect for your work and your production.

-Qingquan

On Thu, Jun 16, 2011 at 7:29 AM, Michael Collins <msc at freeswitch.org> wrote:

> And FTR, this *is* on the wiki - I just didn't know about it. :)
>
> http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation
> (Last item)
>
> -MC
>
>
> On Wed, Jun 15, 2011 at 3:47 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> However, there is not really any rule anywhere that states that bypass
>> media will not touch the packets.
>> All of the packets/sdp are still subject to SIP SOA model so if you
>> pass a remote sdp into the lib, sofia will parse and modify it.
>> The idea of bypass media is to bypass the media by advertising the
>> proper codecs/addrs to arrange point to point media and there is no
>> assurance that the packets will not be modified.
>>
>> However,
>>
>> There is a sofia param called "enable-soa" and a variable
>> sip_enable_soa for per call basis which can be set to false to disable
>> SIP SOA from sofia and most likely result in untouched exchange of
>> SDP.
>>
>> try this in your dialplan
>>
>> <action application="set" data="bypass_media=true"/>
>> <action application="export" data="sip_enable_soa=false"/>
>>
>>
>>
>>
>> On Wed, Jun 15, 2011 at 5:11 PM, Michael Collins <msc at freeswitch.org>
>> wrote:
>> > Ah, quite right. I'd file this in Jira. Hopefully it's not buried deep
>> in
>> > the Sofia stack...
>> > -MC
>> >
>> > On Wed, Jun 15, 2011 at 2:58 PM, Kristian Kielhofner <kris at kriskinc.com
>> >
>> > wrote:
>> >>
>> >> Michael,
>> >>
>> >>  This isn't an SDP offer/answer issue with FreeSWITCH.  Even if there
>> >> were underlying offer/answer issues between the endpoints FreeSWITCH
>> >> is in bypass_media mode and it shouldn't care about the SDP because by
>> >> definition (bypass_media) the SDP is "pass through" between the two
>> >> remote endpoints.
>> >>
>> >>  The specific issue lies in the 183 w/ SDP at line 207.  It has
>> >> rfc2833 payload type 110.  When FreeSWITCH forwards this to the other
>> >> end (line 276) the SDP is untouched (expected with bypass_media)
>> >> EXCEPT the rfc2833 payload type has been changed to 101 (unexpected
>> >> with bypass_media).  It does this again with the 200 OK.  All other
>> >> SDP params are passed through - media address, port, codec, goofy
>> >> session name, and even the SER rtpproxy attributes.  I'll admit the
>> >> remote end/proxy is doing some strange stuff - 100, 101, 183, then
>> >> 180, etc but this does look like a strange bypass_media bug in
>> >> FreeSWITCH.  The RFC2833 payload type should be forwarded between the
>> >> two remote endpoints without being modified by FreeSWITCH - just like
>> >> all of the other SDP parameters (or any part of the SIP body, for that
>> >> matter).
>> >>
>> >> On Wed, Jun 15, 2011 at 5:09 PM, Michael Collins <msc at freeswitch.org>
>> >> wrote:
>> >> > QQ,
>> >> > I don't see that FreeSWITCH is doing anything incorrect here.
>> According
>> >> > to
>> >> > RFC3264, the offerer (FreeSWITCH) sends an SDP and in the case of
>> RTP,
>> >> > the
>> >> > answerer (GRSIP Gateway) must use the payload type offered, even if
>> the
>> >> > answerer uses a different payload type when it sends a
>> telephone-event.
>> >> > http://tools.ietf.org/html/rfc3264#section-6.1
>> >> > Specifically near the end:
>> >> > "In the case of RTP, it MUST use the payload type numbers
>> >> >
>> >> >    from the offer, even if they differ from those in the answer."
>> >> >
>> >> > Technically, FreeSWITCH isn't "changing" anything anyway. The
>> originator
>> >> > of
>> >> > the call (Vox Callcontrol) is the one who chose PT 101 in the INVITE
>> >> > that it
>> >> > sent to FreeSWITCH. FS is just passing that along without modifying
>> it.
>> >> > I think you need to contact the people running the gateway and make
>> sure
>> >> > that they understand that they are not following RFC3264 if they're
>> >> > rejecting telephone-events in PT 101 simply because they prefer to
>> send
>> >> > in
>> >> > PT 110. Also, if your Vox Callcontrol client has any configuration
>> >> > options
>> >> > then maybe you can tell it to use PT 110 for RFC2833 DTMFs. It is a
>> bit
>> >> > of a
>> >> > workaround but this is SIP, so no one is expecting perfection. :)
>> >> > -MC
>> >>
>> >> --
>> >> Kristian Kielhofner
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
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From mays.david at gmail.com  Thu Jun 16 12:32:31 2011
From: mays.david at gmail.com (dma)
Date: Thu, 16 Jun 2011 01:32:31 -0700 (PDT)
Subject: [Freeswitch-users] Leg-A is automatically disconnected on Leg-B
	orginate failure
Message-ID: <1308213151543-6482192.post@n2.nabble.com>

I am creating a call-back solution. After leg-A answers, I originate leg-B
call. After receiving SIP 183 from Leg-B, I bridge the 2 legs. However, in
some cases, leg-A is automatically disconnected by FreeSwitch on leg-B
failure, for example, DESTINATION_OUT_OF_ORDER. The application is not given
a chance to handle leg-B failure event. This should not be a correct
scenario because I never set "hangup-after-bridge", which is false by
default. 

The right way should be, FreeSwitch doesn't hang up leg-A automatically, but
give a chance for the application to decide what to do. 

Please see the logs below:

=================================================

2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
string 0 = [origination_caller_id_number=03996563750914]
2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
string 1 = [originate_timeout=30]
2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
string 2 = [ccd_session_id=20110610105829676824]
2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
string 3 = [sip_cid_type=pid]
2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
string 4 = [privacy=yes]
2011-06-10 11:09:55.370506 [NOTICE] switch_channel.c:808 New Channel
sofia/external/03996590031055 at 203.208.207.212
[ea57b74b-a8c2-4fea-9683-98054dc03a79]
2011-06-10 11:09:55.370506 [DEBUG] mod_sofia.c:4129
(sofia/external/03996590031055 at 203.208.207.212) State Change CS_NEW ->
CS_INIT
2011-06-10 11:09:55.370506 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996590031055 at 203.208.207.212 [BREAK]
2011-06-10 11:09:55.370506 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996590031055 at 203.208.207.212) Running State Change CS_INIT
2011-06-10 11:09:55.370506 [DEBUG] switch_core_state_machine.c:356
(sofia/external/03996590031055 at 203.208.207.212) State INIT
2011-06-10 11:09:55.370506 [DEBUG] mod_sofia.c:84
sofia/external/03996590031055 at 203.208.207.212 SOFIA INIT
send 984 bytes to udp/[203.208.207.212]:5060 at 03:09:55.477997:
   ------------------------------------------------------------------------
   INVITE sip:03996590031055 at 203.208.207.212 SIP/2.0
   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKXjQ7eFpKypy5D
   Max-Forwards: 70
   From: "" &lt;sip:03996563750914 at 202.73.56.46&gt;;tag=H4ZN43UH9S9Qc
   To: &lt;sip:03996590031055 at 203.208.207.212&gt;
   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
   CSeq: 13501633 INVITE
   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, refer
   Privacy: none
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 204
   X-FS-Support: update_display
   P-Asserted-Identity: "" &lt;sip:03996563750914 at 202.73.56.46&gt;
 
   v=0
   o=FreeSWITCH 1307645395 1307645396 IN IP4 202.73.56.46
   s=FreeSWITCH
   c=IN IP4 202.73.56.46
   t=0 0
   m=audio 30000 RTP/AVP 18 3 101 13
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
2011-06-10 11:09:55.371674 [DEBUG] mod_sofia.c:124
(sofia/external/03996590031055 at 203.208.207.212) State Change CS_INIT ->
CS_ROUTING
2011-06-10 11:09:55.371674 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996590031055 at 203.208.207.212 [BREAK]
2011-06-10 11:09:55.371674 [DEBUG] sofia.c:4646 Channel
sofia/external/03996590031055 at 203.208.207.212 entering state [calling][0]
2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:356
(sofia/external/03996590031055 at 203.208.207.212) State INIT going to sleep
2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996590031055 at 203.208.207.212) Running State Change
CS_ROUTING
2011-06-10 11:09:55.373478 [DEBUG] switch_channel.c:1657
(sofia/external/03996590031055 at 203.208.207.212) Callstate Change DOWN ->
RINGING
2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:359
(sofia/external/03996590031055 at 203.208.207.212) State ROUTING
2011-06-10 11:09:55.373478 [DEBUG] mod_sofia.c:147
sofia/external/03996590031055 at 203.208.207.212 SOFIA ROUTING
2011-06-10 11:09:55.373478 [DEBUG] switch_ivr_originate.c:66
(sofia/external/03996590031055 at 203.208.207.212) State Change CS_ROUTING ->
CS_CONSUME_MEDIA
2011-06-10 11:09:55.373478 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996590031055 at 203.208.207.212 [BREAK]
2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:359
(sofia/external/03996590031055 at 203.208.207.212) State ROUTING going to sleep
2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996590031055 at 203.208.207.212) Running State Change
CS_CONSUME_MEDIA
2011-06-10 11:09:55.374485 [DEBUG] switch_core_state_machine.c:378
(sofia/external/03996590031055 at 203.208.207.212) State CONSUME_MEDIA
2011-06-10 11:09:55.374485 [DEBUG] switch_core_state_machine.c:378
(sofia/external/03996590031055 at 203.208.207.212) State CONSUME_MEDIA going to
sleep
recv 307 bytes from udp/[203.208.207.212]:5060 at 03:09:55.485130:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP
10.1.1.46:5080;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
   To: &lt;sip:03996590031055 at 203.208.207.212&gt;
   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
   CSeq: 13501633 INVITE
   Content-Length:     0
 
   ------------------------------------------------------------------------
recv 669 bytes from udp/[203.208.207.212]:5060 at 03:09:56.677788:
   ------------------------------------------------------------------------
   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP
10.1.1.46:5080;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
   To:
&lt;sip:03996590031055 at 203.208.207.212&gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
   CSeq: 13501633 INVITE
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
SUBSCRIBE, UPDATE
   Content-Type: application/sdp
   Content-Length:   189
 
   v=0
   o=- 421265648 1 IN IP4 203.208.207.219
   s=session
   c=IN IP4 203.208.207.196
   t=0 0
   m=audio 30792 RTP/AVP 18 101
   a=rtpmap:18 G729/8000
   a=rtpmap:101 telephone-event/8000
   a=sendrecv
   ------------------------------------------------------------------------
2011-06-10 11:09:56.571839 [INFO] sofia.c:729
sofia/external/03996590031055 at 203.208.207.212 Update Callee ID to "Outbound
Call" <03996590031055>
2011-06-10 11:09:56.571839 [DEBUG] sofia.c:4646 Channel
sofia/external/03996590031055 at 203.208.207.212 entering state
[proceeding][183]
2011-06-10 11:09:56.571839 [DEBUG] sofia.c:4657 Remote SDP:
v=0
o=- 421265648 1 IN IP4 203.208.207.219
s=session
c=IN IP4 203.208.207.196
t=0 0
m=audio 30792 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
 
2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:4467 Audio Codec Compare
[G729:18:8000:20:8000]/[G729:18:8000:20:8000]
2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:2757 Set Codec
sofia/external/03996590031055 at 203.208.207.212 G729/8000 20 ms 160 samples
8000 bits
2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:4565 Set 2833 dtmf send
payload to 101
2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:2987 AUDIO RTP
[sofia/external/03996590031055 at 203.208.207.212] 10.1.1.46 port 30000 ->
203.208.207.196 port 30792 codec: 18 ms: 20
2011-06-10 11:09:56.571839 [DEBUG] switch_rtp.c:1607 Starting timer [soft]
160 bytes per 20ms
2011-06-10 11:09:56.573731 [DEBUG] sofia_glue.c:3228 Set 2833 dtmf send
payload to 101
2011-06-10 11:09:56.573731 [DEBUG] sofia_glue.c:3233 Set 2833 dtmf receive
payload to 101
2011-06-10 11:09:56.573731 [NOTICE] sofia_glue.c:3680 Pre-Answer
sofia/external/03996590031055 at 203.208.207.212!
2011-06-10 11:09:56.573731 [DEBUG] switch_channel.c:2627
(sofia/external/03996590031055 at 203.208.207.212) Callstate Change RINGING ->
EARLY
2011-06-10 11:09:56.573731 [DEBUG] switch_ivr_originate.c:3408 Originate
Resulted in Success: [sofia/external/03996590031055 at 203.208.207.212]
2011-06-10 11:09:56.573731 [DEBUG] mod_commands.c:3205
(sofia/external/03996590031055 at 203.208.207.212) State Change
CS_CONSUME_MEDIA -> CS_EXECUTE
2011-06-10 11:09:56.573731 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996590031055 at 203.208.207.212 [BREAK]
2011-06-10 11:09:56.575262 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996590031055 at 203.208.207.212) Running State Change
CS_EXECUTE
2011-06-10 11:09:56.575262 [DEBUG] switch_core_state_machine.c:366
(sofia/external/03996590031055 at 203.208.207.212) State EXECUTE
2011-06-10 11:09:56.575262 [DEBUG] mod_sofia.c:240
sofia/external/03996590031055 at 203.208.207.212 SOFIA EXECUTE
2011-06-10 11:09:56.575262 [DEBUG] switch_core_state_machine.c:157
sofia/external/03996590031055 at 203.208.207.212 Standard EXECUTE
EXECUTE sofia/external/03996590031055 at 203.208.207.212 park()
2011-06-10 11:09:56.618770 [DEBUG] switch_rtp.c:2933 Correct ip/port
confirmed.
recv 669 bytes from udp/[203.208.207.212]:5060 at 03:10:01.562021:
   ------------------------------------------------------------------------
   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP
10.1.1.46:5080;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
   To:
&lt;sip:03996590031055 at 203.208.207.212&gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
   CSeq: 13501633 INVITE
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
SUBSCRIBE, UPDATE
   Content-Type: application/sdp
   Content-Length:   189
 
   v=0
   o=- 421265648 2 IN IP4 203.208.207.219
   s=session
   c=IN IP4 203.208.207.196
   t=0 0
   m=audio 30792 RTP/AVP 18 101
   a=rtpmap:18 G729/8000
   a=rtpmap:101 telephone-event/8000
   a=sendrecv
   ------------------------------------------------------------------------
2011-06-10 11:10:01.455944 [INFO] sofia.c:729
sofia/external/03996590031055 at 203.208.207.212 Update Callee ID to
"03996590031055" <03996590031055>
2011-06-10 11:10:01.455944 [DEBUG] sofia.c:4641 Channel
sofia/external/03996590031055 at 203.208.207.212 skipping state
[proceeding][183]
recv 669 bytes from udp/[203.208.207.212]:5060 at 03:10:01.563178:
   ------------------------------------------------------------------------
   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP
10.1.1.46:5080;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
   To:
&lt;sip:03996590031055 at 203.208.207.212&gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
   CSeq: 13501633 INVITE
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
SUBSCRIBE, UPDATE
   Content-Type: application/sdp
   Content-Length:   189
 
   v=0
   o=- 421265648 3 IN IP4 203.208.207.219
   s=session
   c=IN IP4 203.208.207.196
   t=0 0
   m=audio 30792 RTP/AVP 18 101
   a=rtpmap:18 G729/8000
   a=rtpmap:101 telephone-event/8000
   a=sendrecv
   ------------------------------------------------------------------------
2011-06-10 11:10:01.457169 [DEBUG] sofia.c:4641 Channel
sofia/external/03996590031055 at 203.208.207.212 skipping state
[proceeding][183]
recv 749 bytes from udp/[203.208.207.212]:5060 at 03:10:27.365284:
   ------------------------------------------------------------------------
   SIP/2.0 200 Ok
   Via: SIP/2.0/UDP
10.1.1.46:5080;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
   To:
&lt;sip:03996590031055 at 203.208.207.212&gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
   CSeq: 13501633 INVITE
   Contact: <sip:203.208.207.212:5060>
   Allow-Events: refer
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
SUBSCRIBE, UPDATE
   Content-Type: application/sdp
   Supported: 100rel, timer, replaces
   Content-Length:   189
 
   v=0
   o=- 421265648 4 IN IP4 203.208.207.219
   s=session
   c=IN IP4 203.208.207.196
   t=0 0
   m=audio 30792 RTP/AVP 18 101
   a=rtpmap:18 G729/8000
   a=rtpmap:101 telephone-event/8000
   a=sendrecv
   ------------------------------------------------------------------------
2011-06-10 11:10:27.258092 [DEBUG] sofia.c:4646 Channel
sofia/external/03996590031055 at 203.208.207.212 entering state
[completing][200]
2011-06-10 11:10:27.258092 [DEBUG] sofia.c:4657 Remote SDP:
v=0
o=- 421265648 4 IN IP4 203.208.207.219
s=session
c=IN IP4 203.208.207.196
t=0 0
m=audio 30792 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
 
send 405 bytes to udp/[203.208.207.212]:5060 at 03:10:27.366562:
   ------------------------------------------------------------------------
   ACK sip:203.208.207.212:5060 SIP/2.0
   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKyUg0ga7pUZmrS
   Max-Forwards: 70
   From: "" &lt;sip:03996563750914 at 202.73.56.46&gt;;tag=H4ZN43UH9S9Qc
   To:
&lt;sip:03996590031055 at 203.208.207.212&gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
   CSeq: 13501633 ACK
   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
   Content-Length: 0
 
   ------------------------------------------------------------------------
2011-06-10 11:10:27.260267 [DEBUG] sofia.c:4646 Channel
sofia/external/03996590031055 at 203.208.207.212 entering state [ready][200]
2011-06-10 11:10:27.260267 [DEBUG] switch_channel.c:2782
(sofia/external/03996590031055 at 203.208.207.212) Callstate Change EARLY ->
ACTIVE
2011-06-10 11:10:27.260267 [NOTICE] sofia.c:5175 Channel
[sofia/external/03996590031055 at 203.208.207.212] has been answered
2011-06-10 11:10:27.264178 [DEBUG] switch_scheduler.c:214 Added task 23
switch_ivr_schedule_hangup (ea57b74b-a8c2-4fea-9683-98054dc03a79) to run at
1307676927
2011-06-10 11:10:27.265202 [DEBUG] switch_core_session.c:954 Send signal
sofia/external/03996590031055 at 203.208.207.212 [BREAK]
2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971 variable
string 0 = [origination_caller_id_number=03996590031055]
2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971 variable
string 1 = [ccd_session_id=20110610105829676824]
2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971 variable
string 2 = [sip_cid_type=pid]
2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971 variable
string 3 = [privacy=yes]
2011-06-10 11:10:27.266218 [NOTICE] switch_channel.c:808 New Channel
sofia/external/03996563750914 at 203.208.207.212
[30228d2b-756a-4a98-871d-db63a2955b52]
2011-06-10 11:10:27.266218 [DEBUG] mod_sofia.c:4129
(sofia/external/03996563750914 at 203.208.207.212) State Change CS_NEW ->
CS_INIT
2011-06-10 11:10:27.266218 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996563750914 at 203.208.207.212 [BREAK]
2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996563750914 at 203.208.207.212) Running State Change CS_INIT
2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:356
(sofia/external/03996563750914 at 203.208.207.212) State INIT
2011-06-10 11:10:27.266218 [DEBUG] mod_sofia.c:84
sofia/external/03996563750914 at 203.208.207.212 SOFIA INIT
2011-06-10 11:10:27.266218 [DEBUG] mod_sofia.c:124
(sofia/external/03996563750914 at 203.208.207.212) State Change CS_INIT ->
CS_ROUTING
2011-06-10 11:10:27.266218 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996563750914 at 203.208.207.212 [BREAK]
2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:356
(sofia/external/03996563750914 at 203.208.207.212) State INIT going to sleep
2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996563750914 at 203.208.207.212) Running State Change
CS_ROUTING
2011-06-10 11:10:27.266218 [DEBUG] switch_channel.c:1657
(sofia/external/03996563750914 at 203.208.207.212) Callstate Change DOWN ->
RINGING
2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:359
(sofia/external/03996563750914 at 203.208.207.212) State ROUTING
send 984 bytes to udp/[203.208.207.212]:5060 at 03:10:27.373220:
   ------------------------------------------------------------------------
   INVITE sip:03996563750914 at 203.208.207.212 SIP/2.0
   Via: SIP/2.0/UDP
202.73.56.46:5080;rport;branch=z9hG4bKZ49rj5Qtr8aBN2011-06-10
11:10:27.266218 [DEBUG] mod_sofia.c:147
sofia/external/03996563750914 at 203.208.207.212 SOFIA ROUTING
 
   Max-Forwards: 70
   From: "" &lt;sip:03996590031055 at 202.73.56.46&gt;;tag=jDSe6ycN62Zar
   To: &lt;sip:03996563750914 at 203.208.207.212&gt;
   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
   CSeq: 13501649 INVITE
   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, refer
   Privacy: none
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 204
   X-FS-Support: update_display
   P-Asserted-Identity: "" &lt;sip:03996590031055 at 202.73.56.46&gt;
 
   v=0
   o=FreeSWITCH 1307646863 1307646864 IN IP4 202.73.56.46
   s=FreeSWITCH
   c=IN IP4 202.73.56.46
   t=0 0
   m=audio 28564 RTP/AVP 18 3 101 13
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
2011-06-10 11:10:27.267630 [DEBUG] sofia.c:4646 Channel
sofia/external/03996563750914 at 203.208.207.212 entering state [calling][0]
2011-06-10 11:10:27.267630 [DEBUG] switch_ivr_originate.c:66
(sofia/external/03996563750914 at 203.208.207.212) State Change CS_ROUTING ->
CS_CONSUME_MEDIA
2011-06-10 11:10:27.267630 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996563750914 at 203.208.207.212 [BREAK]
2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:359
(sofia/external/03996563750914 at 203.208.207.212) State ROUTING going to sleep
2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996563750914 at 203.208.207.212) Running State Change
CS_CONSUME_MEDIA
2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:378
(sofia/external/03996563750914 at 203.208.207.212) State CONSUME_MEDIA
2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:378
(sofia/external/03996563750914 at 203.208.207.212) State CONSUME_MEDIA going to
sleep
recv 307 bytes from udp/[203.208.207.212]:5060 at 03:10:27.378710:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP
10.1.1.46:5080;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
   To: &lt;sip:03996563750914 at 203.208.207.212&gt;
   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
   CSeq: 13501649 INVITE
   Content-Length:     0
 
   ------------------------------------------------------------------------
2011-06-10 11:10:27.278484 [DEBUG] switch_ivr.c:563
sofia/external/03996590031055 at 203.208.207.212 Command Execute
playback(/usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav)
EXECUTE sofia/external/03996590031055 at 203.208.207.212
playback(/usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav)
2011-06-10 11:10:27.278484 [DEBUG] switch_core_file.c:176 File
/usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav sample rate 11025
doesn't match requested rate 8000
2011-06-10 11:10:27.278484 [WARNING] switch_core_file.c:189 File has 2
channels, muxing to mono will occur.
2011-06-10 11:10:27.278484 [DEBUG] switch_ivr_play_say.c:1244 Codec
Activated L16 at 8000hz 2 channels 20ms
2011-06-10 11:10:27.298764 [INFO] mod_com_g729.c:119 ENCODER CREATE -
0x2aaab00310c0 0x2aaab00b20c0
recv 666 bytes from udp/[203.208.207.212]:5060 at 03:10:28.618472:
   ------------------------------------------------------------------------
   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP
10.1.1.46:5080;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
   To:
&lt;sip:03996563750914 at 203.208.207.212&gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
   CSeq: 13501649 INVITE
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
SUBSCRIBE, UPDATE
   Content-Type: application/sdp
   Content-Length:   186
 
   v=0
   o=- 131082 1 IN IP4 203.208.207.218
   s=session
   c=IN IP4 203.208.207.195
   t=0 0
   m=audio 45002 RTP/AVP 18 101
   a=rtpmap:18 G729/8000
   a=rtpmap:101 telephone-event/8000
   a=sendrecv
   ------------------------------------------------------------------------
2011-06-10 11:10:28.513195 [INFO] sofia.c:729
sofia/external/03996563750914 at 203.208.207.212 Update Callee ID to "Outbound
Call" <03996563750914>
2011-06-10 11:10:28.513195 [DEBUG] sofia.c:4646 Channel
sofia/external/03996563750914 at 203.208.207.212 entering state
[proceeding][183]
2011-06-10 11:10:28.513195 [DEBUG] sofia.c:4657 Remote SDP:
v=0
o=- 131082 1 IN IP4 203.208.207.218
s=session
c=IN IP4 203.208.207.195
t=0 0
m=audio 45002 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
 
2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:4467 Audio Codec Compare
[G729:18:8000:20:8000]/[G729:18:8000:20:8000]
2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:2757 Set Codec
sofia/external/03996563750914 at 203.208.207.212 G729/8000 20 ms 160 samples
8000 bits
2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:4565 Set 2833 dtmf send
payload to 101
2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:2987 AUDIO RTP
[sofia/external/03996563750914 at 203.208.207.212] 10.1.1.46 port 28564 ->
203.208.207.195 port 45002 codec: 18 ms: 20
2011-06-10 11:10:28.513195 [DEBUG] switch_rtp.c:1607 Starting timer [soft]
160 bytes per 20ms
2011-06-10 11:10:28.514938 [DEBUG] sofia_glue.c:3228 Set 2833 dtmf send
payload to 101
2011-06-10 11:10:28.514938 [DEBUG] sofia_glue.c:3233 Set 2833 dtmf receive
payload to 101
2011-06-10 11:10:28.514938 [NOTICE] sofia_glue.c:3680 Pre-Answer
sofia/external/03996563750914 at 203.208.207.212!
2011-06-10 11:10:28.514938 [DEBUG] switch_channel.c:2627
(sofia/external/03996563750914 at 203.208.207.212) Callstate Change RINGING ->
EARLY
2011-06-10 11:10:28.514938 [DEBUG] switch_ivr_originate.c:3408 Originate
Resulted in Success: [sofia/external/03996563750914 at 203.208.207.212]
2011-06-10 11:10:28.514938 [DEBUG] mod_commands.c:3205
(sofia/external/03996563750914 at 203.208.207.212) State Change
CS_CONSUME_MEDIA -> CS_EXECUTE
2011-06-10 11:10:28.514938 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996563750914 at 203.208.207.212 [BREAK]
2011-06-10 11:10:28.515884 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996563750914 at 203.208.207.212) Running State Change
CS_EXECUTE
2011-06-10 11:10:28.515884 [DEBUG] switch_core_state_machine.c:366
(sofia/external/03996563750914 at 203.208.207.212) State EXECUTE
2011-06-10 11:10:28.515884 [DEBUG] mod_sofia.c:240
sofia/external/03996563750914 at 203.208.207.212 SOFIA EXECUTE
2011-06-10 11:10:28.515884 [DEBUG] switch_core_state_machine.c:157
sofia/external/03996563750914 at 203.208.207.212 Standard EXECUTE
EXECUTE sofia/external/03996563750914 at 203.208.207.212 park()
2011-06-10 11:10:28.516887 [DEBUG] switch_core_session.c:954 Send signal
sofia/external/03996590031055 at 203.208.207.212 [BREAK]
2011-06-10 11:10:28.517896 [DEBUG] switch_ivr_bridge.c:1480
(sofia/external/03996590031055 at 203.208.207.212) State Change CS_EXECUTE ->
CS_HIBERNATE
2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996590031055 at 203.208.207.212 [BREAK]
2011-06-10 11:10:28.517896 [DEBUG] switch_ivr_bridge.c:1482
(sofia/external/03996563750914 at 203.208.207.212) State Change CS_EXECUTE ->
CS_HIBERNATE
2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996563750914 at 203.208.207.212 [BREAK]
2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:771 Send signal
sofia/external/03996563750914 at 203.208.207.212 [BREAK]
2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:771 Send signal
sofia/external/03996590031055 at 203.208.207.212 [BREAK]
2011-06-10 11:10:28.519223 [DEBUG] switch_ivr_play_say.c:1581 done playing
file
2011-06-10 11:10:28.519223 [DEBUG] switch_ivr.c:563
sofia/external/03996590031055 at 203.208.207.212 Command Execute
playback(tone_stream://%(2000,4000,440,480);loops=10)
2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:366
(sofia/external/03996563750914 at 203.208.207.212) State EXECUTE going to sleep
2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996563750914 at 203.208.207.212) Running State Change
CS_HIBERNATE
2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:381
(sofia/external/03996563750914 at 203.208.207.212) State HIBERNATE
2011-06-10 11:10:28.519223 [DEBUG] mod_sofia.c:221
sofia/external/03996563750914 at 203.208.207.212 SOFIA HIBERNATE
2011-06-10 11:10:28.519223 [DEBUG] switch_ivr_bridge.c:731
(sofia/external/03996563750914 at 203.208.207.212) State Change CS_HIBERNATE ->
CS_RESET
2011-06-10 11:10:28.519223 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996563750914 at 203.208.207.212 [BREAK]
2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:381
(sofia/external/03996563750914 at 203.208.207.212) State HIBERNATE going to
sleep
2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996563750914 at 203.208.207.212) Running State Change
CS_RESET
2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:362
(sofia/external/03996563750914 at 203.208.207.212) State RESET
2011-06-10 11:10:28.519223 [DEBUG] mod_sofia.c:165
sofia/external/03996563750914 at 203.208.207.212 SOFIA RESET
2011-06-10 11:10:28.519223 [DEBUG] switch_ivr_bridge.c:716
sofia/external/03996563750914 at 203.208.207.212 CUSTOM RESET
2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:66
sofia/external/03996563750914 at 203.208.207.212 Standard RESET
2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:362
(sofia/external/03996563750914 at 203.208.207.212) State RESET going to sleep
2011-06-10 11:10:28.519223 [DEBUG] switch_core_session.c:709 Send signal
sofia/external/03996563750914 at 203.208.207.212 [BREAK]
recv 666 bytes from udp/[203.208.207.212]:5060 at 03:10:30.540814:
   ------------------------------------------------------------------------
   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP
10.1.1.46:5080;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
   To:
&lt;sip:03996563750914 at 203.208.207.212&gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
   CSeq: 13501649 INVITE
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
SUBSCRIBE, UPDATE
   Content-Type: application/sdp
   Content-Length:   186
 
   v=0
   o=- 131082 2 IN IP4 203.208.207.218
   s=session
   c=IN IP4 203.208.207.195
   t=0 0
   m=audio 45002 RTP/AVP 18 101
   a=rtpmap:18 G729/8000
   a=rtpmap:101 telephone-event/8000
   a=sendrecv
   ------------------------------------------------------------------------
2011-06-10 11:10:30.435309 [INFO] sofia.c:729
sofia/external/03996563750914 at 203.208.207.212 Update Callee ID to
"03996563750914" <03996563750914>
2011-06-10 11:10:30.435309 [DEBUG] sofia.c:4641 Channel
sofia/external/03996563750914 at 203.208.207.212 skipping state
[proceeding][183]
2011-06-10 11:10:30.691404 [WARNING] switch_core_session.c:1940 Cannot
execute app 'playback' media required on an outbound channel that does not
have media established
2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:366
(sofia/external/03996590031055 at 203.208.207.212) State EXECUTE going to sleep
2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996590031055 at 203.208.207.212) Running State Change
CS_HIBERNATE
2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:381
(sofia/external/03996590031055 at 203.208.207.212) State HIBERNATE
2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:221
sofia/external/03996590031055 at 203.208.207.212 SOFIA HIBERNATE
2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:731
(sofia/external/03996590031055 at 203.208.207.212) State Change CS_HIBERNATE ->
CS_RESET
2011-06-10 11:10:30.691404 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996590031055 at 203.208.207.212 [BREAK]
2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:381
(sofia/external/03996590031055 at 203.208.207.212) State HIBERNATE going to
sleep
2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996590031055 at 203.208.207.212) Running State Change
CS_RESET
2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:362
(sofia/external/03996590031055 at 203.208.207.212) State RESET
2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:165
sofia/external/03996590031055 at 203.208.207.212 SOFIA RESET
2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:716
sofia/external/03996590031055 at 203.208.207.212 CUSTOM RESET
2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:723
(sofia/external/03996590031055 at 203.208.207.212) State Change CS_RESET ->
CS_SOFT_EXECUTE
2011-06-10 11:10:30.691404 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996590031055 at 203.208.207.212 [BREAK]
2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:362
(sofia/external/03996590031055 at 203.208.207.212) State RESET going to sleep
2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996590031055 at 203.208.207.212) Running State Change
CS_SOFT_EXECUTE
2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:372
(sofia/external/03996590031055 at 203.208.207.212) State SOFT_EXECUTE
2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:558 SOFIA SOFT_EXECUTE
2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:741
sofia/external/03996590031055 at 203.208.207.212 CUSTOM SOFT_EXECUTE
2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:761
(sofia/external/03996563750914 at 203.208.207.212) State Change CS_RESET ->
CS_SOFT_EXECUTE
2011-06-10 11:10:30.691404 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996563750914 at 203.208.207.212 [BREAK]
2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996563750914 at 203.208.207.212) Running State Change
CS_SOFT_EXECUTE
2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:372
(sofia/external/03996563750914 at 203.208.207.212) State SOFT_EXECUTE
2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:558 SOFIA SOFT_EXECUTE
2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:741
sofia/external/03996563750914 at 203.208.207.212 CUSTOM SOFT_EXECUTE
2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:204
sofia/external/03996563750914 at 203.208.207.212 Standard SOFT_EXECUTE
2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:372
(sofia/external/03996563750914 at 203.208.207.212) State SOFT_EXECUTE going to
sleep
2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2535
(sofia/external/03996563750914 at 203.208.207.212) Callstate Change EARLY ->
HANGUP
2011-06-10 11:10:30.711798 [NOTICE] switch_ivr_originate.c:1045 Hangup
sofia/external/03996563750914 at 203.208.207.212 [CS_SOFT_EXECUTE]
[DESTINATION_OUT_OF_ORDER]
2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2551 Send signal
sofia/external/03996563750914 at 203.208.207.212 [KILL]
2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996563750914 at 203.208.207.212 [BREAK]
2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996563750914 at 203.208.207.212) Running State Change
CS_HANGUP
2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2535
(sofia/external/03996590031055 at 203.208.207.212) Callstate Change ACTIVE ->
HANGUP
2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
(sofia/external/03996563750914 at 203.208.207.212) State HANGUP
2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:457 Channel
sofia/external/03996563750914 at 203.208.207.212 hanging up, cause:
DESTINATION_OUT_OF_ORDER
2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:510 Sending CANCEL to
sofia/external/03996563750914 at 203.208.207.212
send 390 bytes to udp/[203.208.207.212]:5060 at 03:10:30.818391:
   ------------------------------------------------------------------------
   CANCEL sip:03996563750914 at 203.208.207.212 SIP/2.0
   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKZ49rj5Qtr8aBN
   Max-Forwards: 70
   From: "" &lt;sip:03996590031055 at 202.73.56.46&gt;;tag=jDSe6ycN62Zar
   To: &lt;sip:03996563750914 at 203.208.207.212&gt;
   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
   CSeq: 13501649 CANCEL
   Reason: Q.850;cause=27;text="DESTINATION_OUT_OF_ORDER"
   Content-Length: 0
 
   ------------------------------------------------------------------------
2011-06-10 11:10:30.711798 [NOTICE] switch_ivr_bridge.c:772 Hangup
sofia/external/03996590031055 at 203.208.207.212 [CS_SOFT_EXECUTE]
[ORIGINATOR_CANCEL]
2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:46
sofia/external/03996563750914 at 203.208.207.212 Standard HANGUP, cause:
DESTINATION_OUT_OF_ORDER
2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
(sofia/external/03996563750914 at 203.208.207.212) State HANGUP going to sleep
2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:351
(sofia/external/03996563750914 at 203.208.207.212) State Change CS_HANGUP ->
CS_REPORTING
2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996563750914 at 203.208.207.212 [BREAK]
2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996563750914 at 203.208.207.212) Running State Change
CS_REPORTING
2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:617
(sofia/external/03996563750914 at 203.208.207.212) State REPORTING
2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:53
sofia/external/03996563750914 at 203.208.207.212 Standard REPORTING, cause:
DESTINATION_OUT_OF_ORDER
2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:617
(sofia/external/03996563750914 at 203.208.207.212) State REPORTING going to
sleep
2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2551 Send signal
sofia/external/03996590031055 at 203.208.207.212 [KILL]
2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996590031055 at 203.208.207.212 [BREAK]
2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:372
(sofia/external/03996590031055 at 203.208.207.212) State SOFT_EXECUTE going to
sleep
2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996590031055 at 203.208.207.212) Running State Change
CS_HANGUP
2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:345
(sofia/external/03996563750914 at 203.208.207.212) State Change CS_REPORTING ->
CS_DESTROY
2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996563750914 at 203.208.207.212 [BREAK]
2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1288 Session 40
(sofia/external/03996563750914 at 203.208.207.212) Locked, Waiting on external
entities
2011-06-10 11:10:30.711798 [NOTICE] switch_core_session.c:1306 Session 40
(sofia/external/03996563750914 at 203.208.207.212) Ended
2011-06-10 11:10:30.711798 [NOTICE] switch_core_session.c:1308 Close Channel
sofia/external/03996563750914 at 203.208.207.212 [CS_DESTROY]
2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:449
(sofia/external/03996563750914 at 203.208.207.212) Callstate Change HANGUP ->
DOWN
2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:452
(sofia/external/03996563750914 at 203.208.207.212) Running State Change
CS_DESTROY
2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:462
(sofia/external/03996563750914 at 203.208.207.212) State DESTROY
2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:362
sofia/external/03996563750914 at 203.208.207.212 SOFIA DESTROY
2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
0x2aaaac013028 (nil)
2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
0x2aaaac013028 (nil)
2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
0x2aaaac013088 (nil)
2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
0x2aaaac013088 (nil)
2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:60
sofia/external/03996563750914 at 203.208.207.212 Standard DESTROY
2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:462
(sofia/external/03996563750914 at 203.208.207.212) State DESTROY going to sleep
2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
(sofia/external/03996590031055 at 203.208.207.212) State HANGUP
2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:457 Channel
sofia/external/03996590031055 at 203.208.207.212 hanging up, cause:
ORIGINATOR_CANCEL
2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:500 Sending BYE to
sofia/external/03996590031055 at 203.208.207.212
2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:46
sofia/external/03996590031055 at 203.208.207.212 Standard HANGUP, cause:
ORIGINATOR_CANCEL
2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
(sofia/external/03996590031055 at 203.208.207.212) State HANGUP going to sleep
2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:351
(sofia/external/03996590031055 at 203.208.207.212) State Change CS_HANGUP ->
CS_REPORTING
2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996590031055 at 203.208.207.212 [BREAK]
2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996590031055 at 203.208.207.212) Running State Change
CS_REPORTING
2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:617
(sofia/external/03996590031055 at 203.208.207.212) State REPORTING
2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:53
sofia/external/03996590031055 at 203.208.207.212 Standard REPORTING, cause:
ORIGINATOR_CANCEL
2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:617
(sofia/external/03996590031055 at 203.208.207.212) State REPORTING going to
sleep
2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:345
(sofia/external/03996590031055 at 203.208.207.212) State Change CS_REPORTING ->
CS_DESTROY
2011-06-10 11:10:30.714001 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996590031055 at 203.208.207.212 [BREAK]
2011-06-10 11:10:30.714001 [DEBUG] switch_core_session.c:1288 Session 39
(sofia/external/03996590031055 at 203.208.207.212) Locked, Waiting on external
entities
2011-06-10 11:10:30.714001 [NOTICE] switch_core_session.c:1306 Session 39
(sofia/external/03996590031055 at 203.208.207.212) Ended
2011-06-10 11:10:30.714001 [NOTICE] switch_core_session.c:1308 Close Channel
sofia/external/03996590031055 at 203.208.207.212 [CS_DESTROY]
2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:449
(sofia/external/03996590031055 at 203.208.207.212) Callstate Change HANGUP ->
DOWN
2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:452
(sofia/external/03996590031055 at 203.208.207.212) Running State Change
CS_DESTROY
2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:462
(sofia/external/03996590031055 at 203.208.207.212) State DESTROY
2011-06-10 11:10:30.714001 [DEBUG] mod_sofia.c:362
sofia/external/03996590031055 at 203.208.207.212 SOFIA DESTROY
2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
0x2aaab0031060 (nil)
2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
0x2aaab0031060 (nil)
2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
0x2aaab00310c0 0x2aaab00b20c0
2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
0x2aaab00310c0 (nil)
send 662 bytes to udp/[203.208.207.212]:5060 at 03:10:30.820530:
   ------------------------------------------------------------------------
   BYE sip:203.208.207.212:5060 SIP/2.0
   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bK0D3Hm08XNH1Xg
   Max-Forwards: 70
   From: "" &lt;sip:03996563750914 at 202.73.56.46&gt;;tag=H4ZN43UH9S9Qc
   To:
&lt;sip:03996590031055 at 203.208.207.212&gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
   CSeq: 13501634 BYE
   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL"
   Content-Length: 0
 
   ------------------------------------------------------------------------
2011-06-10 11:10:30.715878 [INFO] mod_com_g729.c:83 ENCODER DESTROY -
0x2aaab00310c0 0x2aaab00b20c0
2011-06-10 11:10:30.715878 [DEBUG] switch_core_state_machine.c:60
sofia/external/03996590031055 at 203.208.207.212 Standard DESTROY
2011-06-10 11:10:30.715878 [DEBUG] switch_core_state_machine.c:462
(sofia/external/03996590031055 at 203.208.207.212) State DESTROY going to sleep
recv 383 bytes from udp/[203.208.207.212]:5060 at 03:10:30.823302:
   ------------------------------------------------------------------------
   SIP/2.0 200 Ok
   Via: SIP/2.0/UDP
10.1.1.46:5080;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
   To:
&lt;sip:03996563750914 at 203.208.207.212&gt;;tag=2QGB951HCR30000E1D00000u00000001QXU3LU
   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
   CSeq: 13501649 CANCEL
   Contact: <sip:203.208.207.212:5060>
   Content-Length:     0
 
   ------------------------------------------------------------------------
recv 411 bytes from udp/[203.208.207.212]:5060 at 03:10:30.824765:
   ------------------------------------------------------------------------
   SIP/2.0 487 Request Terminated
   Via: SIP/2.0/UDP
10.1.1.46:5080;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
   To:
&lt;sip:03996563750914 at 203.208.207.212&gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
   CSeq: 13501649 INVITE
   Reason: SIP;cause=487;text="Request Terminated"
   Content-Length:     0
 
   ------------------------------------------------------------------------
send 371 bytes to udp/[203.208.207.212]:5060 at 03:10:30.824891:
   ------------------------------------------------------------------------
   ACK sip:03996563750914 at 203.208.207.212 SIP/2.0
   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKZ49rj5Qtr8aBN
   Max-Forwards: 70
   From: "" &lt;sip:03996590031055 at 202.73.56.46&gt;;tag=jDSe6ycN62Zar
   To:
&lt;sip:03996563750914 at 203.208.207.212&gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
   CSeq: 13501649 ACK
   Content-Length: 0
 
   ------------------------------------------------------------------------
recv 380 bytes from udp/[203.208.207.212]:5060 at 03:10:30.826375:
   ------------------------------------------------------------------------
   SIP/2.0 200 Ok
   Via: SIP/2.0/UDP
10.1.1.46:5080;rport=5080;branch=z9hG4bK0D3Hm08XNH1Xg;received=10.1.1.46
   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
   To:
&lt;sip:03996590031055 at 203.208.207.212&gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
   CSeq: 13501634 BYE



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Leg-A-is-automatically-disconnected-on-Leg-B-orginate-failure-tp6482192p6482192.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From dave at clancysystems.com  Thu Jun 16 19:54:36 2011
From: dave at clancysystems.com (Dave)
Date: Thu, 16 Jun 2011 09:54:36 -0600
Subject: [Freeswitch-users] Setting Up an Sip Trunk without Authentication
Message-ID: <75EE58D2AC7E4BFE9A53B699375DD699@clancysystems.com>

Hi,

I am new to FreeSWITCH and need to set up FreeSWITCH to receive calls through a Velocity Networks SIP trunk. They say I should not authenticate with Username and Password. Rather, I connect directly to the their IP address.

I put

<extension name="velocity">
      <condition field="destination_number" expression="^2064001950$">
        <action application="bridge" data="sofia/external/$ {destination_number}@their_ip_address"/>
        <action application="lua" data="cme_ivr.lua channel_name"/>
     </condition>
</extension>

in the default.xml under Dialplan

When I Dial the DID from a Phone (not one connected to the FreeSWITCH server) I get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone at theirIP] from their IP, but nothing more.

What I hope to do is take the incoming call from 2064001950 and route it to the Lua IVR script above.

I Appreciate any help you may offer.

Dave Goodwin
 


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From msc at freeswitch.org  Thu Jun 16 22:01:41 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 16 Jun 2011 11:01:41 -0700
Subject: [Freeswitch-users] SRTP
In-Reply-To: <201106161322.53638.justlikeef@gmail.com>
References: <201106152210.41334.justlikeef@gmail.com>
	<BANLkTi==5-N+HgmmU8kHaRq3SpfMAdP6cQ@mail.gmail.com>
	<201106161322.53638.justlikeef@gmail.com>
Message-ID: <BANLkTikchSX+zGw7MfYiLDicdLJ4jo6O+A@mail.gmail.com>

Are you working off of the default.xml dialplan file? If so, it has an
example condition already:

      <condition field="${sip_has_crypto}"
expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$"
break="never">
<action application="set" data="sip_secure_media=true"/>
<!-- Offer SRTP on outbound legs if we have it on inbound. -->
<!-- <action application="export" data="sip_secure_media=true"/> -->
      </condition>

What exactly are you checking on in your scenario? Most likely there is an
elegant way to do it. Give us the plain language description of the problem
you're addressing and the community will no doubt have good suggestions for
you.

-MC

On Thu, Jun 16, 2011 at 10:22 AM, Rob Hutton <justlikeef at gmail.com> wrote:

> Steven -
>
> Thanks for the help here...
>
> So there would have to be two dialplan entries for this number to work with
> either RTP or SRTP? (Maybe two devices registering to the same user?)
>
> Would it make more since to do this in a more global manner higher up in
> the
> dialplan in its own condition block?
>
>
> On Thursday 16 June 2011 03:15:33 Steven Ayre wrote:
> >  {ERR} mod_dialplan_xml.c:110 Nexted conditions are not allowed
> >
> > That's because it shouldn't be nested. It's not missing a /, and the 1st
> > Should have the /. The extra indendation shouldn't be there on the 2nd.
> >
> > It should look like this:
> >
> > <extension name="incoming-fxs">
> >      <condition field="destination_number" expression="^(202)$"/>
> >      <condition field="${sip_has_crypto}"
> > expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$"
> > break="never">
> >           <action application="set" data="sip_secure_media=true"/>
> >           <action application="bridge" data="openzap/1/1"/>
> >      </condition>
> > </extension>
> >
> > The two conditions function as an AND, even though it's not nested. FS
> > stops checking the extension as soon as it sees a condition that's false
> > (at least by default and in the above case), so if the destination is not
> > 202 it'll never get to the 2nd condition.
> >
> > -Steve
> >
> > On 16 June 2011 03:10, Rob Hutton <justlikeef at gmail.com> wrote:
> > > I think I have TLS and SRTP working at this point, but in the docs it
> > > says to use the following template for the dialplan:
> > >
> > > http://wiki.freeswitch.org/wiki/Secure_RTP:
> > >  <extension name="incoming-fxs">
> > >
> > >    <condition field="destination_number" expression="^(202)$"/>
> > >
> > >      <condition field="${sip_has_crypto}"
> > >
> > > expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$"
> > > break="never">
> > >
> > >      <action application="set" data="sip_secure_media=true"/>
> > >      <action application="bridge" data="openzap/1/1"/>
> > >
> > >    </condition>
> > >
> > >  </extension
> > >
> > > 1) There is a missing > at the end of the close extension tag.
> > > 2) There is either a missing / at the end of the internal condition
> line,
> > > or a missing condition close tag somewhere
> > > 3) When I fix the interal condition, I get an error:
> > >
> > > {ERR} mod_dialplan_xml.c:110 Nexted conditions are not allowed
> > >
> > > All this, but a packet capture shows that SRTP is working based on what
> I
> > > did on:
> > >
> > > http://wiki.freeswitch.org/wiki/SIP_TLS
> > >
> > > Can someone give me some guidance on  the Secure_RTP page and I will
> > > update whatever?
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Jun 16 22:15:30 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 16 Jun 2011 11:15:30 -0700
Subject: [Freeswitch-users] Setting Up an Sip Trunk without
	Authentication
In-Reply-To: <75EE58D2AC7E4BFE9A53B699375DD699@clancysystems.com>
References: <75EE58D2AC7E4BFE9A53B699375DD699@clancysystems.com>
Message-ID: <BANLkTinwXNF4zyONuvBOpiDmrXwqYiv=Yw@mail.gmail.com>

Dave,

If you are simply handling incoming calls then you will need to do two
things:

Add Velocity's IP addr to the domains section of acl.conf.xml
Add an extension to the public context (conf/dialplan/public.xml)

Allowing a call into FS via an ACL will send it into the "public" context;
from there you need to transfer it to the default context or just send it
straight to your Lua script.

Also, I don't believe you need to use a "bridge" app based on the
description you gave. Bridge is used to create a new outbound call leg (B
leg) and connect it to the inbound call leg (A leg). If you just are
handling a call with an IVR then there is no B leg needed, thus no need to
bridge.

Welcome to FreeSWITCH, btw! :) I can highly recommend the FS book (the
"bridge" book, ironically) as it discusses some of these basic concepts that
will make your FS experience a whole lot more pleasant.

-MC

On Thu, Jun 16, 2011 at 8:54 AM, Dave <dave at clancysystems.com> wrote:

>  Hi,
>
> I am new to FreeSWITCH and need to set up FreeSWITCH to receive calls
> through a Velocity Networks SIP trunk. They say I should not authenticate
> with Username and Password. Rather, I connect directly to the their IP
> address.
>
> I put
>
> <extension name="velocity">
>       <condition field="destination_number" expression="^2064001950$">
>         <action application="bridge" data="sofia/external/$
> {destination_number}@their_ip_address"/>
>         <action application="lua" data="cme_ivr.lua channel_name"/>
>      </condition>
> </extension>
>
> in the default.xml under Dialplan
>
> When I Dial the DID from a Phone (not one connected to the FreeSWITCH
> server) I get the "SIP auth challenge (INVITE) on sofia profile 'internal'
> for [myphone at theirIP] from their IP, but nothing more.
>
> What I hope to do is take the incoming call from 2064001950 and route it to
> the Lua IVR script above.
>
> I Appreciate any help you may offer.
>
> Dave Goodwin
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Jun 16 22:17:45 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 16 Jun 2011 11:17:45 -0700
Subject: [Freeswitch-users] mod_spandsp event spandsp::rxfaxresult
 variables missing
In-Reply-To: <5003D7D3E06F514E8C682F18D223265C04D3B36C2F@AZWSMS03.azwarranty.int>
References: <5003D7D3E06F514E8C682F18D223265C04D3B36C2B@AZWSMS03.azwarranty.int>
	<BANLkTin3vQii2arQd4KhmSBDoJFivaLiBQ@mail.gmail.com>
	<5003D7D3E06F514E8C682F18D223265C04D3B36C2F@AZWSMS03.azwarranty.int>
Message-ID: <BANLkTikLeff9RL=8mOMvdOjUf998n1aL-A@mail.gmail.com>

On Thu, Jun 16, 2011 at 3:25 AM, Weigel, Stefan <
Stefan.Weigel at allianz-warranty.com> wrote:

>  Hi Steve,
>
>
>
> version shows me only ?git-.
>

This usually means that you have a git version older than 1.7.x and should
probably get a new one. I think we're at least at git 1.7.4...

-MC
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From msc at freeswitch.org  Thu Jun 16 22:21:14 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 16 Jun 2011 11:21:14 -0700
Subject: [Freeswitch-users] Want to play music on hold while executing a
 time consuming operation in Lua
In-Reply-To: <BANLkTin5-d2QSCGCaaOW6j+Y_xu3AfNfog@mail.gmail.com>
References: <BANLkTin5-d2QSCGCaaOW6j+Y_xu3AfNfog@mail.gmail.com>
Message-ID: <BANLkTikVkBbOJLHQSzuQ4FeqphJnFTaSfw@mail.gmail.com>

Actually, I am looking into ways to accomplish these. They are somewhat
related to some recipes I am doing for the FS Cookbook. Please stand by.
FYI, I am looking at an ESL-based solution, not a dialplan script. Thus far
I have not found a viable way to do async stuff like this from a dp script
and I'm thinking ESL is the way to go. I will report back in a bit.

-MC

On Wed, Jun 15, 2011 at 9:53 PM, Sidharth Kshatriya <sid.kshatriya at gmail.com
> wrote:

> I am implementing an IVR using Lua in Freeswitch. In my Lua script I use
> curl to a web service. Sometimes the response takes a long time to come
> back. During that time period I would like to play music on hold.
>
> I have searched the freeswitch discussion archives a lot. There seem to be
> many suggested ways to implement music on hold from a Lua script but the
> answers are not very clear / not really applicable to my use case. I don't
> know what method I should use and which one is recommended.
>
> *Method 1:* Transfer the call to 9664 (music on hold extension). However
> the implementation for this solution for this does not seem to be available
> in Lua. For example: How would I transfer the call back?
> *Method 2:* Using bgapi uuid_park park the call and using uuid_broadcast
> play an audio file. Again what do I do to unpark the call..?
>
> Thanks,
>
> Sidharth
> [P.S. This is a repost, I apologize but I never got any responses on my
> previous email. Need help! :-) ]
>
> --
> Sidharth Kshatriya
> www.sidk.info
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From wes-fs at 499x.com  Thu Jun 16 22:54:52 2011
From: wes-fs at 499x.com (Wes)
Date: Thu, 16 Jun 2011 13:54:52 -0500
Subject: [Freeswitch-users] using keypress to stop a recording.
Message-ID: <4DFA517C.6010109@499x.com>

I'm using the following script to record, and I"m hoping to take a user 
keypress to trigger the end of the recording and then play it back:

(the problem is that it stops on ANY keypress, while it seems like it 
should only break on a "#" keypress)

I found this example at:
http://wiki.freeswitch.org/wiki/Mod_lua#session:recordFile

local numberToCall = 1234
local session = freeswitch.Session("sofia/xxx.xxx.xxx.xxx/"..numberToCall);
session:set_tts_parms("flite", "kal");
session:speak("Thank you for using my recording service, press the pound 
key to stop the recording."  );

function onInput(s, type, obj)
   if (type == "dtmf" and obj['digit'] == '#') then
       return "break";
   end
end

session:setInputCallback("onInput", "");
session:recordFile("/tmp/luatest.wav");
freeswitch.consoleLog("info", "recording stopped by user keypress \n");
session:speak("your voicehas been recorded, i will play it for you now");
session:streamFile("/tmp/luatest.wav");
session:speak("that's it, goodbye.");
freeswitch.consoleLog("info", "hanging up... \n");
session:hangup();



From david.villasmil.work at gmail.com  Thu Jun 16 23:43:53 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 16 Jun 2011 21:43:53 +0200
Subject: [Freeswitch-users] bypass_media problem
In-Reply-To: <BANLkTin8EJuhEgKNov5urDJD-F33BVCQSg@mail.gmail.com>
References: <BANLkTinMGHmdvhrgWXV8OMJadzPSOFcawg@mail.gmail.com>
	<BANLkTin8EJuhEgKNov5urDJD-F33BVCQSg@mail.gmail.com>
Message-ID: <BANLkTi=3ySBjVM+tpXFGeAVr=LwTUS+GRg@mail.gmail.com>

Hello Steve,

my version is:

freeswitch at 192.168.168.3@internal> version
FreeSWITCH Version 1.0.head (git-7e52acf 2011-03-28 22-18-47 -0500)




On Tue, Jun 14, 2011 at 10:14 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Can't see the reason immediately... the channel appears to be hanging up
> because it thinks the bridge is finished.
>
> Can you repeat the debug log, but this time enabling FreeSWITCH's siptrace
> option? (sofia global siptrace on).
>
> Also, I this isn't doing what I think you expect:
>
> session:execute("bridge","{loop=3}sofia/gateway/${distributor(" ..
> route_name .. ")}".. out_number .."")
> The distributor API is called just once to generate the string to give to
> the bridge app, not on each loop. So you'll be dialing through the same
> gateway 3 times not rerouting to another gateway.
>
> Also, what FS version is this?
>
> -Steve
>
>
> On 13 June 2011 23:59, David Villasmil <david.villasmil.work at gmail.com>wrote:
>
>> Hello Guys,
>>
>> I have a set where i'm receiving a call from a gw and sending to another
>> gw. Both un-natted. If I set in my (lua) script bypass_media=false, call
>> goes just fine, if i set it to true, FS sends "Temporary unavailable" to the
>> B-side and CANCELs the A-side...
>>
>> Codec is fine as you will see in the pastebin..
>>
>> rou
>> here's the trace:
>>
>> http://pastebin.freeswitch.org/16480
>>
>> and here's the console log:
>>
>> http://pastebin.freeswitch.org/16481
>>
>> I'm bridging with a lua script, this is the pertinent part:
>>
>> gw_sip_username and gw_sip_pwd are just bogus "1234","1234", no
>> registration is needed....
>>
>>
>>
>>
>>
>>  session:execute("set","continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,407")
>>
>>      session:execute("set","inbound-late-negotiation=true")
>>      session:execute("set","inbound-bypass-media=false")
>>
>>      session:execute("set","proxy_media=false")
>>      session:execute("set","bypass_media=false")
>>
>>
>>      --session:execute("bridge","{sip_auth_username=" .. gw_sip_username
>> .. ",sip_auth_password=" .. gw_sip_pwd .. "}sofia/external/".. out_number
>> .."@".. gw_sip_ip .."")
>>
>>
>>      fsLog("BRIDGE EXECUTE:", "{loop=3}sofia/gateway/${distributor(" ..
>> route_name .. ")}" .. out_number .. "")
>>      session:execute("bridge","{loop=3}sofia/gateway/${distributor(" ..
>> route_name .. ")}".. out_number .."")
>>
>>      -- hangup
>>      session:hangup();
>>
>>
>>
>> Thanks guys for your help!
>>
>>
>> David
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Thu Jun 16 23:49:53 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 16 Jun 2011 20:49:53 +0100
Subject: [Freeswitch-users] bypass_media problem
In-Reply-To: <BANLkTi=3ySBjVM+tpXFGeAVr=LwTUS+GRg@mail.gmail.com>
References: <BANLkTinMGHmdvhrgWXV8OMJadzPSOFcawg@mail.gmail.com>
	<BANLkTin8EJuhEgKNov5urDJD-F33BVCQSg@mail.gmail.com>
	<BANLkTi=3ySBjVM+tpXFGeAVr=LwTUS+GRg@mail.gmail.com>
Message-ID: <BANLkTi=0tt76GHqqvtEBb8tMrdtHnX8pog@mail.gmail.com>

Ok. Have you tried reproducing it on the latest Git?

-Steve


On 16 June 2011 20:43, David Villasmil <david.villasmil.work at gmail.com>wrote:

> Hello Steve,
>
> my version is:
>
> freeswitch at 192.168.168.3@internal> version
> FreeSWITCH Version 1.0.head (git-7e52acf 2011-03-28 22-18-47 -0500)
>
>
>
>
> On Tue, Jun 14, 2011 at 10:14 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Can't see the reason immediately... the channel appears to be hanging up
>> because it thinks the bridge is finished.
>>
>> Can you repeat the debug log, but this time enabling FreeSWITCH's siptrace
>> option? (sofia global siptrace on).
>>
>> Also, I this isn't doing what I think you expect:
>>
>> session:execute("bridge","{loop=3}sofia/gateway/${distributor(" ..
>> route_name .. ")}".. out_number .."")
>> The distributor API is called just once to generate the string to give to
>> the bridge app, not on each loop. So you'll be dialing through the same
>> gateway 3 times not rerouting to another gateway.
>>
>> Also, what FS version is this?
>>
>> -Steve
>>
>>
>> On 13 June 2011 23:59, David Villasmil <david.villasmil.work at gmail.com>wrote:
>>
>>> Hello Guys,
>>>
>>> I have a set where i'm receiving a call from a gw and sending to another
>>> gw. Both un-natted. If I set in my (lua) script bypass_media=false, call
>>> goes just fine, if i set it to true, FS sends "Temporary unavailable" to the
>>> B-side and CANCELs the A-side...
>>>
>>> Codec is fine as you will see in the pastebin..
>>>
>>> rou
>>> here's the trace:
>>>
>>> http://pastebin.freeswitch.org/16480
>>>
>>> and here's the console log:
>>>
>>> http://pastebin.freeswitch.org/16481
>>>
>>> I'm bridging with a lua script, this is the pertinent part:
>>>
>>> gw_sip_username and gw_sip_pwd are just bogus "1234","1234", no
>>> registration is needed....
>>>
>>>
>>>
>>>
>>>
>>>  session:execute("set","continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,407")
>>>
>>>      session:execute("set","inbound-late-negotiation=true")
>>>      session:execute("set","inbound-bypass-media=false")
>>>
>>>      session:execute("set","proxy_media=false")
>>>      session:execute("set","bypass_media=false")
>>>
>>>
>>>      --session:execute("bridge","{sip_auth_username=" .. gw_sip_username
>>> .. ",sip_auth_password=" .. gw_sip_pwd .. "}sofia/external/".. out_number
>>> .."@".. gw_sip_ip .."")
>>>
>>>
>>>      fsLog("BRIDGE EXECUTE:", "{loop=3}sofia/gateway/${distributor(" ..
>>> route_name .. ")}" .. out_number .. "")
>>>      session:execute("bridge","{loop=3}sofia/gateway/${distributor(" ..
>>> route_name .. ")}".. out_number .."")
>>>
>>>      -- hangup
>>>      session:hangup();
>>>
>>>
>>>
>>> Thanks guys for your help!
>>>
>>>
>>> David
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dave at clancysystems.com  Fri Jun 17 00:02:14 2011
From: dave at clancysystems.com (Dave)
Date: Thu, 16 Jun 2011 14:02:14 -0600
Subject: [Freeswitch-users] Setting Up an Sip Trunk without
	Authentication
References: <75EE58D2AC7E4BFE9A53B699375DD699@clancysystems.com>
	<BANLkTinwXNF4zyONuvBOpiDmrXwqYiv=Yw@mail.gmail.com>
Message-ID: <5A9A76E6D31143F1B2C971E2A7508D22@clancysystems.com>


Under "domains" in acl.conf.xml I added  

   <node type="allow" value="xxx.xxx.xxx.xxx"/>

x's are the IP address it's just a single address not a CIDR.


And I created the file velocity_did.xml with the following code..

<include>
  <extension name="velocity_did">
    <condition field="destination_number" expression="^(2064001950)$">
      <action application="set" data="domain_name=$${domain}"/>
      <action application="answer"/>
   <action application="lua" data="cme_ivr.lua channel_name"/>
 
 </condition>
  </extension>
</include>

I still just get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone at theirIP] from their IP, but nothing more.

Do I need to change the line ..

<action application="set" data="domain_name=$${domain}"/>

to the IP address? If so what syntax?

They have no Domain Name.

Thanks again for your help.

Dave Goodwin      

  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Thursday, June 16, 2011 12:15 PM
  Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication


  Dave,


  If you are simply handling incoming calls then you will need to do two things:


  Add Velocity's IP addr to the domains section of acl.conf.xml
  Add an extension to the public context (conf/dialplan/public.xml)


  Allowing a call into FS via an ACL will send it into the "public" context; from there you need to transfer it to the default context or just send it straight to your Lua script.


  Also, I don't believe you need to use a "bridge" app based on the description you gave. Bridge is used to create a new outbound call leg (B leg) and connect it to the inbound call leg (A leg). If you just are handling a call with an IVR then there is no B leg needed, thus no need to bridge.


  Welcome to FreeSWITCH, btw! :) I can highly recommend the FS book (the "bridge" book, ironically) as it discusses some of these basic concepts that will make your FS experience a whole lot more pleasant.


  -MC


  On Thu, Jun 16, 2011 at 8:54 AM, Dave <dave at clancysystems.com> wrote:

    Hi,

    I am new to FreeSWITCH and need to set up FreeSWITCH to receive calls through a Velocity Networks SIP trunk. They say I should not authenticate with Username and Password. Rather, I connect directly to the their IP address.

    I put

    <extension name="velocity">
          <condition field="destination_number" expression="^2064001950$">
            <action application="bridge" data="sofia/external/$ {destination_number}@their_ip_address"/>
            <action application="lua" data="cme_ivr.lua channel_name"/>
         </condition>
    </extension>

    in the default.xml under Dialplan

    When I Dial the DID from a Phone (not one connected to the FreeSWITCH server) I get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone at theirIP] from their IP, but nothing more.

    What I hope to do is take the incoming call from 2064001950 and route it to the Lua IVR script above.

    I Appreciate any help you may offer.

    Dave Goodwin





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



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From msc at freeswitch.org  Fri Jun 17 00:02:03 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 16 Jun 2011 13:02:03 -0700
Subject: [Freeswitch-users] Test message - Please ignore
Message-ID: <BANLkTimdjZLmP3R_64vTEt2eiryf3hbYUQ@mail.gmail.com>

I <3 FS
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From msc at freeswitch.org  Fri Jun 17 00:11:04 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 16 Jun 2011 13:11:04 -0700
Subject: [Freeswitch-users] Setting Up an Sip Trunk without
	Authentication
In-Reply-To: <5A9A76E6D31143F1B2C971E2A7508D22@clancysystems.com>
References: <75EE58D2AC7E4BFE9A53B699375DD699@clancysystems.com>
	<BANLkTinwXNF4zyONuvBOpiDmrXwqYiv=Yw@mail.gmail.com>
	<5A9A76E6D31143F1B2C971E2A7508D22@clancysystems.com>
Message-ID: <BANLkTimg6hEv0Vbg_+WkjKcuADp5dz0hNA@mail.gmail.com>

On Thu, Jun 16, 2011 at 1:02 PM, Dave <dave at clancysystems.com> wrote:

>
> Under "domains" in acl.conf.xml I added
>
>    <node type="allow" value="xxx.xxx.xxx.xxx"/>
>
>
The correct syntax is:
<node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>


> x's are the IP address it's just a single address not a CIDR.
>
>
> And I created the file velocity_did.xml with the following code..
>
> <include>
>   <extension name="velocity_did">
>
>     <condition field="destination_number" expression="^(2064001950)$">
>       <action application="set" data="domain_name=$${domain}"/>
>       <action application="answer"/>
>
>    <action application="lua" data="cme_ivr.lua channel_name"/>
>
>  </condition>
>   </extension>
> </include>
> I still just get the "SIP auth challenge (INVITE) on sofia profile
> 'internal' for [myphone at theirIP] from their IP, but nothing more.
>
>
Wait - can you explain *exactly* what you're doing for testing? Are you
trying to dial out via Velocity and back in to your DID? If so then you have
2 completely different things you need to set up. The instructions I gave
were only for inbound DID, so use a cell phone or something to test that.

For outbound it seems like they are sending you an auth challenge, which
means they need to give you a username and password as well as a host/ip.
You need to create a gateway, preferably in your external profile. Just be
sure to set the "register" param to false since they are not expecting you
to register with them.

-MC

Do I need to change the line ..
>
> <action application="set" data="domain_name=$${domain}"/>
> to the IP address? If so what syntax?
>
> They have no Domain Name.
>
> Thanks again for your help.
>
> Dave Goodwin
>
>
> ----- Original Message -----
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Thursday, June 16, 2011 12:15 PM
> *Subject:* Re: [Freeswitch-users] Setting Up an Sip Trunk without
> Authentication
>
> Dave,
>
> If you are simply handling incoming calls then you will need to do two
> things:
>
> Add Velocity's IP addr to the domains section of acl.conf.xml
> Add an extension to the public context (conf/dialplan/public.xml)
>
> Allowing a call into FS via an ACL will send it into the "public" context;
> from there you need to transfer it to the default context or just send it
> straight to your Lua script.
>
> Also, I don't believe you need to use a "bridge" app based on the
> description you gave. Bridge is used to create a new outbound call leg (B
> leg) and connect it to the inbound call leg (A leg). If you just are
> handling a call with an IVR then there is no B leg needed, thus no need to
> bridge.
>
> Welcome to FreeSWITCH, btw! :) I can highly recommend the FS book (the
> "bridge" book, ironically) as it discusses some of these basic concepts that
> will make your FS experience a whole lot more pleasant.
>
> -MC
>
> On Thu, Jun 16, 2011 at 8:54 AM, Dave <dave at clancysystems.com> wrote:
>
>>  Hi,
>>
>> I am new to FreeSWITCH and need to set up FreeSWITCH to receive calls
>> through a Velocity Networks SIP trunk. They say I should not authenticate
>> with Username and Password. Rather, I connect directly to the their IP
>> address.
>>
>> I put
>>
>> <extension name="velocity">
>>       <condition field="destination_number" expression="^2064001950$">
>>         <action application="bridge" data="sofia/external/$
>> {destination_number}@their_ip_address"/>
>>         <action application="lua" data="cme_ivr.lua channel_name"/>
>>      </condition>
>> </extension>
>>
>> in the default.xml under Dialplan
>>
>> When I Dial the DID from a Phone (not one connected to the FreeSWITCH
>> server) I get the "SIP auth challenge (INVITE) on sofia profile 'internal'
>> for [myphone at theirIP] from their IP, but nothing more.
>>
>> What I hope to do is take the incoming call from 2064001950 and route it
>> to the Lua IVR script above.
>>
>> I Appreciate any help you may offer.
>>
>> Dave Goodwin
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dave at clancysystems.com  Fri Jun 17 00:27:32 2011
From: dave at clancysystems.com (Dave)
Date: Thu, 16 Jun 2011 14:27:32 -0600
Subject: [Freeswitch-users] Setting Up an Sip Trunk without
	Authentication
References: <75EE58D2AC7E4BFE9A53B699375DD699@clancysystems.com><BANLkTinwXNF4zyONuvBOpiDmrXwqYiv=Yw@mail.gmail.com><5A9A76E6D31143F1B2C971E2A7508D22@clancysystems.com>
	<BANLkTimg6hEv0Vbg_+WkjKcuADp5dz0hNA@mail.gmail.com>
Message-ID: <AF706099B82D4F309EDBEA0ACE7DA5D4@clancysystems.com>

Yes, I am only testing inbound.  When I call from our office phone (not connected to FreeSWITCH, it's on a trunk from broadvox)  I Get a busy signal and at that moment FreeSWITCH shows...
"SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone at theirIP] from their IP.

If I call from my cell phone I get the Operator Message "your call did not go through" and at that moment the same thing shows in FreeSWITCH
"SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone at theirIP] from their IP.
 

  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Thursday, June 16, 2011 2:11 PM
  Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication





  On Thu, Jun 16, 2011 at 1:02 PM, Dave <dave at clancysystems.com> wrote:


    Under "domains" in acl.conf.xml I added  

       <node type="allow" value="xxx.xxx.xxx.xxx"/>

  The correct syntax is:
  <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>

    x's are the IP address it's just a single address not a CIDR.


    And I created the file velocity_did.xml with the following code..

    <include>
      <extension name="velocity_did">

        <condition field="destination_number" expression="^(2064001950)$">

          <action application="set" data="domain_name=$${domain}"/>
          <action application="answer"/>

       <action application="lua" data="cme_ivr.lua channel_name"/>
     
     </condition>
      </extension>

    </include>

    I still just get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone at theirIP] from their IP, but nothing more.

  Wait - can you explain *exactly* what you're doing for testing? Are you trying to dial out via Velocity and back in to your DID? If so then you have 2 completely different things you need to set up. The instructions I gave were only for inbound DID, so use a cell phone or something to test that.


  For outbound it seems like they are sending you an auth challenge, which means they need to give you a username and password as well as a host/ip. You need to create a gateway, preferably in your external profile. Just be sure to set the "register" param to false since they are not expecting you to register with them. 


  -MC


    Do I need to change the line ..

    <action application="set" data="domain_name=$${domain}"/>

    to the IP address? If so what syntax?

    They have no Domain Name.

    Thanks again for your help.

    Dave Goodwin      

      ----- Original Message ----- 
      From: Michael Collins 
      To: FreeSWITCH Users Help 
      Sent: Thursday, June 16, 2011 12:15 PM
      Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication


      Dave, 


      If you are simply handling incoming calls then you will need to do two things:


      Add Velocity's IP addr to the domains section of acl.conf.xml
      Add an extension to the public context (conf/dialplan/public.xml)


      Allowing a call into FS via an ACL will send it into the "public" context; from there you need to transfer it to the default context or just send it straight to your Lua script.


      Also, I don't believe you need to use a "bridge" app based on the description you gave. Bridge is used to create a new outbound call leg (B leg) and connect it to the inbound call leg (A leg). If you just are handling a call with an IVR then there is no B leg needed, thus no need to bridge.


      Welcome to FreeSWITCH, btw! :) I can highly recommend the FS book (the "bridge" book, ironically) as it discusses some of these basic concepts that will make your FS experience a whole lot more pleasant.


      -MC


      On Thu, Jun 16, 2011 at 8:54 AM, Dave <dave at clancysystems.com> wrote:

        Hi,

        I am new to FreeSWITCH and need to set up FreeSWITCH to receive calls through a Velocity Networks SIP trunk. They say I should not authenticate with Username and Password. Rather, I connect directly to the their IP address.

        I put

        <extension name="velocity">
              <condition field="destination_number" expression="^2064001950$">
                <action application="bridge" data="sofia/external/$ {destination_number}@their_ip_address"/>
                <action application="lua" data="cme_ivr.lua channel_name"/>
             </condition>
        </extension>

        in the default.xml under Dialplan

        When I Dial the DID from a Phone (not one connected to the FreeSWITCH server) I get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone at theirIP] from their IP, but nothing more.

        What I hope to do is take the incoming call from 2064001950 and route it to the Lua IVR script above.

        I Appreciate any help you may offer.

        Dave Goodwin





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





    _______________________________________________
    Join us at ClueCon 2011, Aug 9-11, Chicago
    http://www.cluecon.com 877-7-4ACLUE
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org



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From msc at freeswitch.org  Fri Jun 17 00:46:58 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 16 Jun 2011 13:46:58 -0700
Subject: [Freeswitch-users] Setting Up an Sip Trunk without
	Authentication
In-Reply-To: <AF706099B82D4F309EDBEA0ACE7DA5D4@clancysystems.com>
References: <75EE58D2AC7E4BFE9A53B699375DD699@clancysystems.com>
	<BANLkTinwXNF4zyONuvBOpiDmrXwqYiv=Yw@mail.gmail.com>
	<5A9A76E6D31143F1B2C971E2A7508D22@clancysystems.com>
	<BANLkTimg6hEv0Vbg_+WkjKcuADp5dz0hNA@mail.gmail.com>
	<AF706099B82D4F309EDBEA0ACE7DA5D4@clancysystems.com>
Message-ID: <BANLkTimR8-x4yWk=QFymZymVVK2JdPt_-g@mail.gmail.com>

Did you reloadxml and restart the sofia profile after the changes? Also, you
need to "reloadacl" after doing a change to acl.conf.xml.

-MC

On Thu, Jun 16, 2011 at 1:27 PM, Dave <dave at clancysystems.com> wrote:

>  Yes, I am only testing inbound.  When I call from our office phone (not
> connected to FreeSWITCH, it's on a trunk from broadvox)  I Get a busy signal
> and at that moment FreeSWITCH shows...
>  "SIP auth challenge (INVITE) on sofia profile 'internal' for
> [myphone at theirIP] from their IP.
>
> If I call from my cell phone I get the Operator Message "your call did not
> go through" and at that moment the same thing shows in FreeSWITCH
>  "SIP auth challenge (INVITE) on sofia profile 'internal' for
> [myphone at theirIP] from their IP.
>
>
>
> ----- Original Message -----
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Thursday, June 16, 2011 2:11 PM
> *Subject:* Re: [Freeswitch-users] Setting Up an Sip Trunk without
> Authentication
>
>
>
> On Thu, Jun 16, 2011 at 1:02 PM, Dave <dave at clancysystems.com> wrote:
>
>>
>> Under "domains" in acl.conf.xml I added
>>
>>    <node type="allow" value="xxx.xxx.xxx.xxx"/>
>>
>>
> The correct syntax is:
> <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>
>
>
>>  x's are the IP address it's just a single address not a CIDR.
>>
>>
>> And I created the file velocity_did.xml with the following code..
>>
>> <include>
>>   <extension name="velocity_did">
>>
>>     <condition field="destination_number" expression="^(2064001950)$">
>>       <action application="set" data="domain_name=$${domain}"/>
>>       <action application="answer"/>
>>
>>    <action application="lua" data="cme_ivr.lua channel_name"/>
>>
>>  </condition>
>>   </extension>
>> </include>
>> I still just get the "SIP auth challenge (INVITE) on sofia profile
>> 'internal' for [myphone at theirIP] from their IP, but nothing more.
>>
>>
> Wait - can you explain *exactly* what you're doing for testing? Are you
> trying to dial out via Velocity and back in to your DID? If so then you have
> 2 completely different things you need to set up. The instructions I gave
> were only for inbound DID, so use a cell phone or something to test that.
>
> For outbound it seems like they are sending you an auth challenge, which
> means they need to give you a username and password as well as a host/ip.
> You need to create a gateway, preferably in your external profile. Just be
> sure to set the "register" param to false since they are not expecting you
> to register with them.
>
> -MC
>
>  Do I need to change the line ..
>>
>> <action application="set" data="domain_name=$${domain}"/>
>> to the IP address? If so what syntax?
>>
>> They have no Domain Name.
>>
>> Thanks again for your help.
>>
>> Dave Goodwin
>>
>>
>> ----- Original Message -----
>> *From:* Michael Collins <msc at freeswitch.org>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Sent:* Thursday, June 16, 2011 12:15 PM
>> *Subject:* Re: [Freeswitch-users] Setting Up an Sip Trunk without
>> Authentication
>>
>> Dave,
>>
>> If you are simply handling incoming calls then you will need to do two
>> things:
>>
>> Add Velocity's IP addr to the domains section of acl.conf.xml
>> Add an extension to the public context (conf/dialplan/public.xml)
>>
>> Allowing a call into FS via an ACL will send it into the "public" context;
>> from there you need to transfer it to the default context or just send it
>> straight to your Lua script.
>>
>> Also, I don't believe you need to use a "bridge" app based on the
>> description you gave. Bridge is used to create a new outbound call leg (B
>> leg) and connect it to the inbound call leg (A leg). If you just are
>> handling a call with an IVR then there is no B leg needed, thus no need to
>> bridge.
>>
>> Welcome to FreeSWITCH, btw! :) I can highly recommend the FS book (the
>> "bridge" book, ironically) as it discusses some of these basic concepts that
>> will make your FS experience a whole lot more pleasant.
>>
>> -MC
>>
>> On Thu, Jun 16, 2011 at 8:54 AM, Dave <dave at clancysystems.com> wrote:
>>
>>>  Hi,
>>>
>>> I am new to FreeSWITCH and need to set up FreeSWITCH to receive calls
>>> through a Velocity Networks SIP trunk. They say I should not authenticate
>>> with Username and Password. Rather, I connect directly to the their IP
>>> address.
>>>
>>> I put
>>>
>>> <extension name="velocity">
>>>       <condition field="destination_number" expression="^2064001950$">
>>>         <action application="bridge" data="sofia/external/$
>>> {destination_number}@their_ip_address"/>
>>>         <action application="lua" data="cme_ivr.lua channel_name"/>
>>>      </condition>
>>> </extension>
>>>
>>> in the default.xml under Dialplan
>>>
>>> When I Dial the DID from a Phone (not one connected to the FreeSWITCH
>>> server) I get the "SIP auth challenge (INVITE) on sofia profile 'internal'
>>> for [myphone at theirIP] from their IP, but nothing more.
>>>
>>> What I hope to do is take the incoming call from 2064001950 and route it
>>> to the Lua IVR script above.
>>>
>>> I Appreciate any help you may offer.
>>>
>>> Dave Goodwin
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From grsingh750 at gmail.com  Fri Jun 17 00:50:57 2011
From: grsingh750 at gmail.com (guru singh)
Date: Fri, 17 Jun 2011 02:20:57 +0530
Subject: [Freeswitch-users] Making a external call to Mobile number
In-Reply-To: <BANLkTinFuhpjGkLv14UC3LGUYcFc8H9cjQ@mail.gmail.com>
References: <BANLkTinFuhpjGkLv14UC3LGUYcFc8H9cjQ@mail.gmail.com>
Message-ID: <BANLkTik7dWYP8e5pjLVQkk2V0hHVUN4+jA@mail.gmail.com>

Hi Ankit,

Look at the debug level logs in fs_cli and let us know what they say.
It'll be more relevant than "this call is not connecting". It should
tell you exactly why its failing, also if things dont work always look
here first instead of what x-lite is telling you.

Regards
guru



On Thu, Jun 16, 2011 at 10:34 PM, ankIT WALiA <ankitwalia4u at gmail.com> wrote:
> Dear all,
>
> I am trying to make an external call to a mobile number of India using the
> default ext number 1005. For doing this, I did the following things.
>
> First, I created a user at IPTEL.org, I added a SIP profile. I registered
> with IPTEL. I also added a dial plan regex for 91 - (10 digit number). But,
> because of some issue, I was not able to connect to IPTEL SIP. I was getting
> error on X-LITE about some server issue.
>
> then, I tried my VOIP account with ActionVoip which is a paid account and
> has SIP service. I could connect to my action voip account through X-LITE
> SIP.
>
> I made a SIP profile for Action Voip and registered the username with action
> voip in FS. On reload, it got registered using action voip SIP profile. I
> changed the dialplan to use actionvoip gateway.
>
> But still, when I am trying to call to my mobile number in India. The call
> is not able to connect.
>
> I think there is gap in my knowledge. Please help me I am new to FreeSwitch.
>
> Thanks
> Ankit
>
> --
> Life is like a rose its upto u feel it as its fragrance or thorns
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Fri Jun 17 00:51:22 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 16 Jun 2011 13:51:22 -0700
Subject: [Freeswitch-users] Leg-A is automatically disconnected on Leg-B
 orginate failure
In-Reply-To: <1308213151543-6482192.post@n2.nabble.com>
References: <1308213151543-6482192.post@n2.nabble.com>
Message-ID: <BANLkTi=xYgBTPVGxVGmk9f2=bmX5X2vF=Q@mail.gmail.com>

How about setting this?
http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail

-MC

On Thu, Jun 16, 2011 at 1:32 AM, dma <mays.david at gmail.com> wrote:

> I am creating a call-back solution. After leg-A answers, I originate leg-B
> call. After receiving SIP 183 from Leg-B, I bridge the 2 legs. However, in
> some cases, leg-A is automatically disconnected by FreeSwitch on leg-B
> failure, for example, DESTINATION_OUT_OF_ORDER. The application is not
> given
> a chance to handle leg-B failure event. This should not be a correct
> scenario because I never set "hangup-after-bridge", which is false by
> default.
>
> The right way should be, FreeSwitch doesn't hang up leg-A automatically,
> but
> give a chance for the application to decide what to do.
>
> Please see the logs below:
>
> =================================================
>
> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
> string 0 = [origination_caller_id_number=03996563750914]
> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
> string 1 = [originate_timeout=30]
> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
> string 2 = [ccd_session_id=20110610105829676824]
> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
> string 3 = [sip_cid_type=pid]
> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
> string 4 = [privacy=yes]
> 2011-06-10 11:09:55.370506 [NOTICE] switch_channel.c:808 New Channel
> sofia/external/03996590031055 at 203.208.207.212
> [ea57b74b-a8c2-4fea-9683-98054dc03a79]
> 2011-06-10 11:09:55.370506 [DEBUG] mod_sofia.c:4129
> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_NEW ->
> CS_INIT
> 2011-06-10 11:09:55.370506 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
> 2011-06-10 11:09:55.370506 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
> CS_INIT
> 2011-06-10 11:09:55.370506 [DEBUG] switch_core_state_machine.c:356
> (sofia/external/03996590031055 at 203.208.207.212) State INIT
> 2011-06-10 11:09:55.370506 [DEBUG] mod_sofia.c:84
> sofia/external/03996590031055 at 203.208.207.212 SOFIA INIT
> send 984 bytes to udp/[203.208.207.212]:5060 at 03:09:55.477997:
>   ------------------------------------------------------------------------
>   INVITE sip:03996590031055 at 203.208.207.212 SIP/2.0
>   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKXjQ7eFpKypy5D
>   Max-Forwards: 70
>   From: "" &lt;sip:03996563750914 at 202.73.56.46&gt;;tag=H4ZN43UH9S9Qc
>   To: &lt;sip:03996590031055 at 203.208.207.212&gt;
>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>   CSeq: 13501633 INVITE
>   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
>   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
>   Supported: timer, precondition, path, replaces
>   Allow-Events: talk, hold, refer
>   Privacy: none
>   Content-Type: application/sdp
>   Content-Disposition: session
>   Content-Length: 204
>   X-FS-Support: update_display
>   P-Asserted-Identity: "" &lt;sip:03996563750914 at 202.73.56.46&gt;
>
>   v=0
>   o=FreeSWITCH 1307645395 1307645396 IN IP4 202.73.56.46
>   s=FreeSWITCH
>   c=IN IP4 202.73.56.46
>   t=0 0
>   m=audio 30000 RTP/AVP 18 3 101 13
>   a=rtpmap:101 telephone-event/8000
>   a=fmtp:101 0-16
>   a=ptime:20
>   ------------------------------------------------------------------------
> 2011-06-10 11:09:55.371674 [DEBUG] mod_sofia.c:124
> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_INIT ->
> CS_ROUTING
> 2011-06-10 11:09:55.371674 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
> 2011-06-10 11:09:55.371674 [DEBUG] sofia.c:4646 Channel
> sofia/external/03996590031055 at 203.208.207.212 entering state [calling][0]
> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:356
> (sofia/external/03996590031055 at 203.208.207.212) State INIT going to sleep
> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
> CS_ROUTING
> 2011-06-10 11:09:55.373478 [DEBUG] switch_channel.c:1657
> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change DOWN ->
> RINGING
> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:359
> (sofia/external/03996590031055 at 203.208.207.212) State ROUTING
> 2011-06-10 11:09:55.373478 [DEBUG] mod_sofia.c:147
> sofia/external/03996590031055 at 203.208.207.212 SOFIA ROUTING
> 2011-06-10 11:09:55.373478 [DEBUG] switch_ivr_originate.c:66
> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_ROUTING ->
> CS_CONSUME_MEDIA
> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:359
> (sofia/external/03996590031055 at 203.208.207.212) State ROUTING going to
> sleep
> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
> CS_CONSUME_MEDIA
> 2011-06-10 11:09:55.374485 [DEBUG] switch_core_state_machine.c:378
> (sofia/external/03996590031055 at 203.208.207.212) State CONSUME_MEDIA
> 2011-06-10 11:09:55.374485 [DEBUG] switch_core_state_machine.c:378
> (sofia/external/03996590031055 at 203.208.207.212) State CONSUME_MEDIA going
> to
> sleep
> recv 307 bytes from udp/[203.208.207.212]:5060 at 03:09:55.485130:
>   ------------------------------------------------------------------------
>   SIP/2.0 100 Trying
>   Via: SIP/2.0/UDP
> 10.1.1.46:5080;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>   To: &lt;sip:03996590031055 at 203.208.207.212&gt;
>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>   CSeq: 13501633 INVITE
>   Content-Length:     0
>
>   ------------------------------------------------------------------------
> recv 669 bytes from udp/[203.208.207.212]:5060 at 03:09:56.677788:
>   ------------------------------------------------------------------------
>   SIP/2.0 183 Session Progress
>   Via: SIP/2.0/UDP
> 10.1.1.46:5080;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>   To:
> &lt;sip:03996590031055 at 203.208.207.212
> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>   CSeq: 13501633 INVITE
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
> SUBSCRIBE, UPDATE
>   Content-Type: application/sdp
>   Content-Length:   189
>
>   v=0
>   o=- 421265648 1 IN IP4 203.208.207.219
>   s=session
>   c=IN IP4 203.208.207.196
>   t=0 0
>   m=audio 30792 RTP/AVP 18 101
>   a=rtpmap:18 G729/8000
>   a=rtpmap:101 telephone-event/8000
>   a=sendrecv
>   ------------------------------------------------------------------------
> 2011-06-10 11:09:56.571839 [INFO] sofia.c:729
> sofia/external/03996590031055 at 203.208.207.212 Update Callee ID to
> "Outbound
> Call" <03996590031055>
> 2011-06-10 11:09:56.571839 [DEBUG] sofia.c:4646 Channel
> sofia/external/03996590031055 at 203.208.207.212 entering state
> [proceeding][183]
> 2011-06-10 11:09:56.571839 [DEBUG] sofia.c:4657 Remote SDP:
> v=0
> o=- 421265648 1 IN IP4 203.208.207.219
> s=session
> c=IN IP4 203.208.207.196
> t=0 0
> m=audio 30792 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=rtpmap:101 telephone-event/8000
>
> 2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:4467 Audio Codec Compare
> [G729:18:8000:20:8000]/[G729:18:8000:20:8000]
> 2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:2757 Set Codec
> sofia/external/03996590031055 at 203.208.207.212 G729/8000 20 ms 160 samples
> 8000 bits
> 2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:4565 Set 2833 dtmf send
> payload to 101
> 2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:2987 AUDIO RTP
> [sofia/external/03996590031055 at 203.208.207.212] 10.1.1.46 port 30000 ->
> 203.208.207.196 port 30792 codec: 18 ms: 20
> 2011-06-10 11:09:56.571839 [DEBUG] switch_rtp.c:1607 Starting timer [soft]
> 160 bytes per 20ms
> 2011-06-10 11:09:56.573731 [DEBUG] sofia_glue.c:3228 Set 2833 dtmf send
> payload to 101
> 2011-06-10 11:09:56.573731 [DEBUG] sofia_glue.c:3233 Set 2833 dtmf receive
> payload to 101
> 2011-06-10 11:09:56.573731 [NOTICE] sofia_glue.c:3680 Pre-Answer
> sofia/external/03996590031055 at 203.208.207.212!
> 2011-06-10 11:09:56.573731 [DEBUG] switch_channel.c:2627
> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change RINGING
> ->
> EARLY
> 2011-06-10 11:09:56.573731 [DEBUG] switch_ivr_originate.c:3408 Originate
> Resulted in Success: [sofia/external/03996590031055 at 203.208.207.212]
> 2011-06-10 11:09:56.573731 [DEBUG] mod_commands.c:3205
> (sofia/external/03996590031055 at 203.208.207.212) State Change
> CS_CONSUME_MEDIA -> CS_EXECUTE
> 2011-06-10 11:09:56.573731 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
> 2011-06-10 11:09:56.575262 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
> CS_EXECUTE
> 2011-06-10 11:09:56.575262 [DEBUG] switch_core_state_machine.c:366
> (sofia/external/03996590031055 at 203.208.207.212) State EXECUTE
> 2011-06-10 11:09:56.575262 [DEBUG] mod_sofia.c:240
> sofia/external/03996590031055 at 203.208.207.212 SOFIA EXECUTE
> 2011-06-10 11:09:56.575262 [DEBUG] switch_core_state_machine.c:157
> sofia/external/03996590031055 at 203.208.207.212 Standard EXECUTE
> EXECUTE sofia/external/03996590031055 at 203.208.207.212 park()
> 2011-06-10 11:09:56.618770 [DEBUG] switch_rtp.c:2933 Correct ip/port
> confirmed.
> recv 669 bytes from udp/[203.208.207.212]:5060 at 03:10:01.562021:
>   ------------------------------------------------------------------------
>   SIP/2.0 183 Session Progress
>   Via: SIP/2.0/UDP
> 10.1.1.46:5080;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>   To:
> &lt;sip:03996590031055 at 203.208.207.212
> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>   CSeq: 13501633 INVITE
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
> SUBSCRIBE, UPDATE
>   Content-Type: application/sdp
>   Content-Length:   189
>
>   v=0
>   o=- 421265648 2 IN IP4 203.208.207.219
>   s=session
>   c=IN IP4 203.208.207.196
>   t=0 0
>   m=audio 30792 RTP/AVP 18 101
>   a=rtpmap:18 G729/8000
>   a=rtpmap:101 telephone-event/8000
>   a=sendrecv
>   ------------------------------------------------------------------------
> 2011-06-10 11:10:01.455944 [INFO] sofia.c:729
> sofia/external/03996590031055 at 203.208.207.212 Update Callee ID to
> "03996590031055" <03996590031055>
> 2011-06-10 11:10:01.455944 [DEBUG] sofia.c:4641 Channel
> sofia/external/03996590031055 at 203.208.207.212 skipping state
> [proceeding][183]
> recv 669 bytes from udp/[203.208.207.212]:5060 at 03:10:01.563178:
>   ------------------------------------------------------------------------
>   SIP/2.0 183 Session Progress
>   Via: SIP/2.0/UDP
> 10.1.1.46:5080;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>   To:
> &lt;sip:03996590031055 at 203.208.207.212
> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>   CSeq: 13501633 INVITE
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
> SUBSCRIBE, UPDATE
>   Content-Type: application/sdp
>   Content-Length:   189
>
>   v=0
>   o=- 421265648 3 IN IP4 203.208.207.219
>   s=session
>   c=IN IP4 203.208.207.196
>   t=0 0
>   m=audio 30792 RTP/AVP 18 101
>   a=rtpmap:18 G729/8000
>   a=rtpmap:101 telephone-event/8000
>   a=sendrecv
>   ------------------------------------------------------------------------
> 2011-06-10 11:10:01.457169 [DEBUG] sofia.c:4641 Channel
> sofia/external/03996590031055 at 203.208.207.212 skipping state
> [proceeding][183]
> recv 749 bytes from udp/[203.208.207.212]:5060 at 03:10:27.365284:
>   ------------------------------------------------------------------------
>   SIP/2.0 200 Ok
>   Via: SIP/2.0/UDP
> 10.1.1.46:5080;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>   To:
> &lt;sip:03996590031055 at 203.208.207.212
> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>   CSeq: 13501633 INVITE
>   Contact: <sip:203.208.207.212:5060>
>   Allow-Events: refer
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
> SUBSCRIBE, UPDATE
>   Content-Type: application/sdp
>   Supported: 100rel, timer, replaces
>   Content-Length:   189
>
>   v=0
>   o=- 421265648 4 IN IP4 203.208.207.219
>   s=session
>   c=IN IP4 203.208.207.196
>   t=0 0
>   m=audio 30792 RTP/AVP 18 101
>   a=rtpmap:18 G729/8000
>   a=rtpmap:101 telephone-event/8000
>   a=sendrecv
>   ------------------------------------------------------------------------
> 2011-06-10 11:10:27.258092 [DEBUG] sofia.c:4646 Channel
> sofia/external/03996590031055 at 203.208.207.212 entering state
> [completing][200]
> 2011-06-10 11:10:27.258092 [DEBUG] sofia.c:4657 Remote SDP:
> v=0
> o=- 421265648 4 IN IP4 203.208.207.219
> s=session
> c=IN IP4 203.208.207.196
> t=0 0
> m=audio 30792 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=rtpmap:101 telephone-event/8000
>
> send 405 bytes to udp/[203.208.207.212]:5060 at 03:10:27.366562:
>   ------------------------------------------------------------------------
>   ACK sip:203.208.207.212:5060 SIP/2.0
>   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKyUg0ga7pUZmrS
>   Max-Forwards: 70
>   From: "" &lt;sip:03996563750914 at 202.73.56.46&gt;;tag=H4ZN43UH9S9Qc
>   To:
> &lt;sip:03996590031055 at 203.208.207.212
> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>   CSeq: 13501633 ACK
>   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
>   Content-Length: 0
>
>   ------------------------------------------------------------------------
> 2011-06-10 11:10:27.260267 [DEBUG] sofia.c:4646 Channel
> sofia/external/03996590031055 at 203.208.207.212 entering state [ready][200]
> 2011-06-10 11:10:27.260267 [DEBUG] switch_channel.c:2782
> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change EARLY ->
> ACTIVE
> 2011-06-10 11:10:27.260267 [NOTICE] sofia.c:5175 Channel
> [sofia/external/03996590031055 at 203.208.207.212] has been answered
> 2011-06-10 11:10:27.264178 [DEBUG] switch_scheduler.c:214 Added task 23
> switch_ivr_schedule_hangup (ea57b74b-a8c2-4fea-9683-98054dc03a79) to run at
> 1307676927
> 2011-06-10 11:10:27.265202 [DEBUG] switch_core_session.c:954 Send signal
> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
> 2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971 variable
> string 0 = [origination_caller_id_number=03996590031055]
> 2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971 variable
> string 1 = [ccd_session_id=20110610105829676824]
> 2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971 variable
> string 2 = [sip_cid_type=pid]
> 2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971 variable
> string 3 = [privacy=yes]
> 2011-06-10 11:10:27.266218 [NOTICE] switch_channel.c:808 New Channel
> sofia/external/03996563750914 at 203.208.207.212
> [30228d2b-756a-4a98-871d-db63a2955b52]
> 2011-06-10 11:10:27.266218 [DEBUG] mod_sofia.c:4129
> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_NEW ->
> CS_INIT
> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
> CS_INIT
> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:356
> (sofia/external/03996563750914 at 203.208.207.212) State INIT
> 2011-06-10 11:10:27.266218 [DEBUG] mod_sofia.c:84
> sofia/external/03996563750914 at 203.208.207.212 SOFIA INIT
> 2011-06-10 11:10:27.266218 [DEBUG] mod_sofia.c:124
> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_INIT ->
> CS_ROUTING
> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:356
> (sofia/external/03996563750914 at 203.208.207.212) State INIT going to sleep
> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
> CS_ROUTING
> 2011-06-10 11:10:27.266218 [DEBUG] switch_channel.c:1657
> (sofia/external/03996563750914 at 203.208.207.212) Callstate Change DOWN ->
> RINGING
> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:359
> (sofia/external/03996563750914 at 203.208.207.212) State ROUTING
> send 984 bytes to udp/[203.208.207.212]:5060 at 03:10:27.373220:
>   ------------------------------------------------------------------------
>   INVITE sip:03996563750914 at 203.208.207.212 SIP/2.0
>   Via: SIP/2.0/UDP
> 202.73.56.46:5080;rport;branch=z9hG4bKZ49rj5Qtr8aBN2011-06-10
> 11:10:27.266218 [DEBUG] mod_sofia.c:147
> sofia/external/03996563750914 at 203.208.207.212 SOFIA ROUTING
>
>   Max-Forwards: 70
>   From: "" &lt;sip:03996590031055 at 202.73.56.46&gt;;tag=jDSe6ycN62Zar
>   To: &lt;sip:03996563750914 at 203.208.207.212&gt;
>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>   CSeq: 13501649 INVITE
>   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
>   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
>   Supported: timer, precondition, path, replaces
>   Allow-Events: talk, hold, refer
>   Privacy: none
>   Content-Type: application/sdp
>   Content-Disposition: session
>   Content-Length: 204
>   X-FS-Support: update_display
>   P-Asserted-Identity: "" &lt;sip:03996590031055 at 202.73.56.46&gt;
>
>   v=0
>   o=FreeSWITCH 1307646863 1307646864 IN IP4 202.73.56.46
>   s=FreeSWITCH
>   c=IN IP4 202.73.56.46
>   t=0 0
>   m=audio 28564 RTP/AVP 18 3 101 13
>   a=rtpmap:101 telephone-event/8000
>   a=fmtp:101 0-16
>   a=ptime:20
>   ------------------------------------------------------------------------
> 2011-06-10 11:10:27.267630 [DEBUG] sofia.c:4646 Channel
> sofia/external/03996563750914 at 203.208.207.212 entering state [calling][0]
> 2011-06-10 11:10:27.267630 [DEBUG] switch_ivr_originate.c:66
> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_ROUTING ->
> CS_CONSUME_MEDIA
> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:359
> (sofia/external/03996563750914 at 203.208.207.212) State ROUTING going to
> sleep
> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
> CS_CONSUME_MEDIA
> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:378
> (sofia/external/03996563750914 at 203.208.207.212) State CONSUME_MEDIA
> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:378
> (sofia/external/03996563750914 at 203.208.207.212) State CONSUME_MEDIA going
> to
> sleep
> recv 307 bytes from udp/[203.208.207.212]:5060 at 03:10:27.378710:
>   ------------------------------------------------------------------------
>   SIP/2.0 100 Trying
>   Via: SIP/2.0/UDP
> 10.1.1.46:5080;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>   To: &lt;sip:03996563750914 at 203.208.207.212&gt;
>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>   CSeq: 13501649 INVITE
>   Content-Length:     0
>
>   ------------------------------------------------------------------------
> 2011-06-10 11:10:27.278484 [DEBUG] switch_ivr.c:563
> sofia/external/03996590031055 at 203.208.207.212 Command Execute
> playback(/usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav)
> EXECUTE sofia/external/03996590031055 at 203.208.207.212
> playback(/usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav)
> 2011-06-10 11:10:27.278484 [DEBUG] switch_core_file.c:176 File
> /usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav sample rate
> 11025
> doesn't match requested rate 8000
> 2011-06-10 11:10:27.278484 [WARNING] switch_core_file.c:189 File has 2
> channels, muxing to mono will occur.
> 2011-06-10 11:10:27.278484 [DEBUG] switch_ivr_play_say.c:1244 Codec
> Activated L16 at 8000hz 2 channels 20ms
> 2011-06-10 11:10:27.298764 [INFO] mod_com_g729.c:119 ENCODER CREATE -
> 0x2aaab00310c0 0x2aaab00b20c0
> recv 666 bytes from udp/[203.208.207.212]:5060 at 03:10:28.618472:
>   ------------------------------------------------------------------------
>   SIP/2.0 183 Session Progress
>   Via: SIP/2.0/UDP
> 10.1.1.46:5080;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>   To:
> &lt;sip:03996563750914 at 203.208.207.212
> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>   CSeq: 13501649 INVITE
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
> SUBSCRIBE, UPDATE
>   Content-Type: application/sdp
>   Content-Length:   186
>
>   v=0
>   o=- 131082 1 IN IP4 203.208.207.218
>   s=session
>   c=IN IP4 203.208.207.195
>   t=0 0
>   m=audio 45002 RTP/AVP 18 101
>   a=rtpmap:18 G729/8000
>   a=rtpmap:101 telephone-event/8000
>   a=sendrecv
>   ------------------------------------------------------------------------
> 2011-06-10 11:10:28.513195 [INFO] sofia.c:729
> sofia/external/03996563750914 at 203.208.207.212 Update Callee ID to
> "Outbound
> Call" <03996563750914>
> 2011-06-10 11:10:28.513195 [DEBUG] sofia.c:4646 Channel
> sofia/external/03996563750914 at 203.208.207.212 entering state
> [proceeding][183]
> 2011-06-10 11:10:28.513195 [DEBUG] sofia.c:4657 Remote SDP:
> v=0
> o=- 131082 1 IN IP4 203.208.207.218
> s=session
> c=IN IP4 203.208.207.195
> t=0 0
> m=audio 45002 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=rtpmap:101 telephone-event/8000
>
> 2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:4467 Audio Codec Compare
> [G729:18:8000:20:8000]/[G729:18:8000:20:8000]
> 2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:2757 Set Codec
> sofia/external/03996563750914 at 203.208.207.212 G729/8000 20 ms 160 samples
> 8000 bits
> 2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:4565 Set 2833 dtmf send
> payload to 101
> 2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:2987 AUDIO RTP
> [sofia/external/03996563750914 at 203.208.207.212] 10.1.1.46 port 28564 ->
> 203.208.207.195 port 45002 codec: 18 ms: 20
> 2011-06-10 11:10:28.513195 [DEBUG] switch_rtp.c:1607 Starting timer [soft]
> 160 bytes per 20ms
> 2011-06-10 11:10:28.514938 [DEBUG] sofia_glue.c:3228 Set 2833 dtmf send
> payload to 101
> 2011-06-10 11:10:28.514938 [DEBUG] sofia_glue.c:3233 Set 2833 dtmf receive
> payload to 101
> 2011-06-10 11:10:28.514938 [NOTICE] sofia_glue.c:3680 Pre-Answer
> sofia/external/03996563750914 at 203.208.207.212!
> 2011-06-10 11:10:28.514938 [DEBUG] switch_channel.c:2627
> (sofia/external/03996563750914 at 203.208.207.212) Callstate Change RINGING
> ->
> EARLY
> 2011-06-10 11:10:28.514938 [DEBUG] switch_ivr_originate.c:3408 Originate
> Resulted in Success: [sofia/external/03996563750914 at 203.208.207.212]
> 2011-06-10 11:10:28.514938 [DEBUG] mod_commands.c:3205
> (sofia/external/03996563750914 at 203.208.207.212) State Change
> CS_CONSUME_MEDIA -> CS_EXECUTE
> 2011-06-10 11:10:28.514938 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
> 2011-06-10 11:10:28.515884 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
> CS_EXECUTE
> 2011-06-10 11:10:28.515884 [DEBUG] switch_core_state_machine.c:366
> (sofia/external/03996563750914 at 203.208.207.212) State EXECUTE
> 2011-06-10 11:10:28.515884 [DEBUG] mod_sofia.c:240
> sofia/external/03996563750914 at 203.208.207.212 SOFIA EXECUTE
> 2011-06-10 11:10:28.515884 [DEBUG] switch_core_state_machine.c:157
> sofia/external/03996563750914 at 203.208.207.212 Standard EXECUTE
> EXECUTE sofia/external/03996563750914 at 203.208.207.212 park()
> 2011-06-10 11:10:28.516887 [DEBUG] switch_core_session.c:954 Send signal
> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
> 2011-06-10 11:10:28.517896 [DEBUG] switch_ivr_bridge.c:1480
> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_EXECUTE ->
> CS_HIBERNATE
> 2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
> 2011-06-10 11:10:28.517896 [DEBUG] switch_ivr_bridge.c:1482
> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_EXECUTE ->
> CS_HIBERNATE
> 2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
> 2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:771 Send signal
> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
> 2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:771 Send signal
> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
> 2011-06-10 11:10:28.519223 [DEBUG] switch_ivr_play_say.c:1581 done playing
> file
> 2011-06-10 11:10:28.519223 [DEBUG] switch_ivr.c:563
> sofia/external/03996590031055 at 203.208.207.212 Command Execute
> playback(tone_stream://%(2000,4000,440,480);loops=10)
> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:366
> (sofia/external/03996563750914 at 203.208.207.212) State EXECUTE going to
> sleep
> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
> CS_HIBERNATE
> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:381
> (sofia/external/03996563750914 at 203.208.207.212) State HIBERNATE
> 2011-06-10 11:10:28.519223 [DEBUG] mod_sofia.c:221
> sofia/external/03996563750914 at 203.208.207.212 SOFIA HIBERNATE
> 2011-06-10 11:10:28.519223 [DEBUG] switch_ivr_bridge.c:731
> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_HIBERNATE
> ->
> CS_RESET
> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:381
> (sofia/external/03996563750914 at 203.208.207.212) State HIBERNATE going to
> sleep
> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
> CS_RESET
> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:362
> (sofia/external/03996563750914 at 203.208.207.212) State RESET
> 2011-06-10 11:10:28.519223 [DEBUG] mod_sofia.c:165
> sofia/external/03996563750914 at 203.208.207.212 SOFIA RESET
> 2011-06-10 11:10:28.519223 [DEBUG] switch_ivr_bridge.c:716
> sofia/external/03996563750914 at 203.208.207.212 CUSTOM RESET
> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:66
> sofia/external/03996563750914 at 203.208.207.212 Standard RESET
> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:362
> (sofia/external/03996563750914 at 203.208.207.212) State RESET going to sleep
> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_session.c:709 Send signal
> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
> recv 666 bytes from udp/[203.208.207.212]:5060 at 03:10:30.540814:
>   ------------------------------------------------------------------------
>   SIP/2.0 183 Session Progress
>   Via: SIP/2.0/UDP
> 10.1.1.46:5080;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>   To:
> &lt;sip:03996563750914 at 203.208.207.212
> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>   CSeq: 13501649 INVITE
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
> SUBSCRIBE, UPDATE
>   Content-Type: application/sdp
>   Content-Length:   186
>
>   v=0
>   o=- 131082 2 IN IP4 203.208.207.218
>   s=session
>   c=IN IP4 203.208.207.195
>   t=0 0
>   m=audio 45002 RTP/AVP 18 101
>   a=rtpmap:18 G729/8000
>   a=rtpmap:101 telephone-event/8000
>   a=sendrecv
>   ------------------------------------------------------------------------
> 2011-06-10 11:10:30.435309 [INFO] sofia.c:729
> sofia/external/03996563750914 at 203.208.207.212 Update Callee ID to
> "03996563750914" <03996563750914>
> 2011-06-10 11:10:30.435309 [DEBUG] sofia.c:4641 Channel
> sofia/external/03996563750914 at 203.208.207.212 skipping state
> [proceeding][183]
> 2011-06-10 11:10:30.691404 [WARNING] switch_core_session.c:1940 Cannot
> execute app 'playback' media required on an outbound channel that does not
> have media established
> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:366
> (sofia/external/03996590031055 at 203.208.207.212) State EXECUTE going to
> sleep
> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
> CS_HIBERNATE
> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:381
> (sofia/external/03996590031055 at 203.208.207.212) State HIBERNATE
> 2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:221
> sofia/external/03996590031055 at 203.208.207.212 SOFIA HIBERNATE
> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:731
> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_HIBERNATE
> ->
> CS_RESET
> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:381
> (sofia/external/03996590031055 at 203.208.207.212) State HIBERNATE going to
> sleep
> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
> CS_RESET
> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:362
> (sofia/external/03996590031055 at 203.208.207.212) State RESET
> 2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:165
> sofia/external/03996590031055 at 203.208.207.212 SOFIA RESET
> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:716
> sofia/external/03996590031055 at 203.208.207.212 CUSTOM RESET
> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:723
> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_RESET ->
> CS_SOFT_EXECUTE
> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:362
> (sofia/external/03996590031055 at 203.208.207.212) State RESET going to sleep
> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
> CS_SOFT_EXECUTE
> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:372
> (sofia/external/03996590031055 at 203.208.207.212) State SOFT_EXECUTE
> 2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:558 SOFIA SOFT_EXECUTE
> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:741
> sofia/external/03996590031055 at 203.208.207.212 CUSTOM SOFT_EXECUTE
> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:761
> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_RESET ->
> CS_SOFT_EXECUTE
> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
> CS_SOFT_EXECUTE
> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:372
> (sofia/external/03996563750914 at 203.208.207.212) State SOFT_EXECUTE
> 2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:558 SOFIA SOFT_EXECUTE
> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:741
> sofia/external/03996563750914 at 203.208.207.212 CUSTOM SOFT_EXECUTE
> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:204
> sofia/external/03996563750914 at 203.208.207.212 Standard SOFT_EXECUTE
> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:372
> (sofia/external/03996563750914 at 203.208.207.212) State SOFT_EXECUTE going
> to
> sleep
> 2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2535
> (sofia/external/03996563750914 at 203.208.207.212) Callstate Change EARLY ->
> HANGUP
> 2011-06-10 11:10:30.711798 [NOTICE] switch_ivr_originate.c:1045 Hangup
> sofia/external/03996563750914 at 203.208.207.212 [CS_SOFT_EXECUTE]
> [DESTINATION_OUT_OF_ORDER]
> 2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2551 Send signal
> sofia/external/03996563750914 at 203.208.207.212 [KILL]
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
> CS_HANGUP
> 2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2535
> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change ACTIVE ->
> HANGUP
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
> (sofia/external/03996563750914 at 203.208.207.212) State HANGUP
> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:457 Channel
> sofia/external/03996563750914 at 203.208.207.212 hanging up, cause:
> DESTINATION_OUT_OF_ORDER
> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:510 Sending CANCEL to
> sofia/external/03996563750914 at 203.208.207.212
> send 390 bytes to udp/[203.208.207.212]:5060 at 03:10:30.818391:
>   ------------------------------------------------------------------------
>   CANCEL sip:03996563750914 at 203.208.207.212 SIP/2.0
>   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKZ49rj5Qtr8aBN
>   Max-Forwards: 70
>   From: "" &lt;sip:03996590031055 at 202.73.56.46&gt;;tag=jDSe6ycN62Zar
>   To: &lt;sip:03996563750914 at 203.208.207.212&gt;
>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>   CSeq: 13501649 CANCEL
>   Reason: Q.850;cause=27;text="DESTINATION_OUT_OF_ORDER"
>   Content-Length: 0
>
>   ------------------------------------------------------------------------
> 2011-06-10 11:10:30.711798 [NOTICE] switch_ivr_bridge.c:772 Hangup
> sofia/external/03996590031055 at 203.208.207.212 [CS_SOFT_EXECUTE]
> [ORIGINATOR_CANCEL]
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:46
> sofia/external/03996563750914 at 203.208.207.212 Standard HANGUP, cause:
> DESTINATION_OUT_OF_ORDER
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
> (sofia/external/03996563750914 at 203.208.207.212) State HANGUP going to
> sleep
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:351
> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_HANGUP ->
> CS_REPORTING
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
> CS_REPORTING
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:617
> (sofia/external/03996563750914 at 203.208.207.212) State REPORTING
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:53
> sofia/external/03996563750914 at 203.208.207.212 Standard REPORTING, cause:
> DESTINATION_OUT_OF_ORDER
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:617
> (sofia/external/03996563750914 at 203.208.207.212) State REPORTING going to
> sleep
> 2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2551 Send signal
> sofia/external/03996590031055 at 203.208.207.212 [KILL]
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:372
> (sofia/external/03996590031055 at 203.208.207.212) State SOFT_EXECUTE going
> to
> sleep
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
> CS_HANGUP
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:345
> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_REPORTING
> ->
> CS_DESTROY
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1288 Session 40
> (sofia/external/03996563750914 at 203.208.207.212) Locked, Waiting on
> external
> entities
> 2011-06-10 11:10:30.711798 [NOTICE] switch_core_session.c:1306 Session 40
> (sofia/external/03996563750914 at 203.208.207.212) Ended
> 2011-06-10 11:10:30.711798 [NOTICE] switch_core_session.c:1308 Close
> Channel
> sofia/external/03996563750914 at 203.208.207.212 [CS_DESTROY]
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:449
> (sofia/external/03996563750914 at 203.208.207.212) Callstate Change HANGUP ->
> DOWN
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:452
> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
> CS_DESTROY
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:462
> (sofia/external/03996563750914 at 203.208.207.212) State DESTROY
> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:362
> sofia/external/03996563750914 at 203.208.207.212 SOFIA DESTROY
> 2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> 0x2aaaac013028 (nil)
> 2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
> 0x2aaaac013028 (nil)
> 2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> 0x2aaaac013088 (nil)
> 2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
> 0x2aaaac013088 (nil)
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:60
> sofia/external/03996563750914 at 203.208.207.212 Standard DESTROY
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:462
> (sofia/external/03996563750914 at 203.208.207.212) State DESTROY going to
> sleep
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
> (sofia/external/03996590031055 at 203.208.207.212) State HANGUP
> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:457 Channel
> sofia/external/03996590031055 at 203.208.207.212 hanging up, cause:
> ORIGINATOR_CANCEL
> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:500 Sending BYE to
> sofia/external/03996590031055 at 203.208.207.212
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:46
> sofia/external/03996590031055 at 203.208.207.212 Standard HANGUP, cause:
> ORIGINATOR_CANCEL
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
> (sofia/external/03996590031055 at 203.208.207.212) State HANGUP going to
> sleep
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:351
> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_HANGUP ->
> CS_REPORTING
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
> CS_REPORTING
> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:617
> (sofia/external/03996590031055 at 203.208.207.212) State REPORTING
> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:53
> sofia/external/03996590031055 at 203.208.207.212 Standard REPORTING, cause:
> ORIGINATOR_CANCEL
> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:617
> (sofia/external/03996590031055 at 203.208.207.212) State REPORTING going to
> sleep
> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:345
> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_REPORTING
> ->
> CS_DESTROY
> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_session.c:1288 Session 39
> (sofia/external/03996590031055 at 203.208.207.212) Locked, Waiting on
> external
> entities
> 2011-06-10 11:10:30.714001 [NOTICE] switch_core_session.c:1306 Session 39
> (sofia/external/03996590031055 at 203.208.207.212) Ended
> 2011-06-10 11:10:30.714001 [NOTICE] switch_core_session.c:1308 Close
> Channel
> sofia/external/03996590031055 at 203.208.207.212 [CS_DESTROY]
> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:449
> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change HANGUP ->
> DOWN
> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:452
> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
> CS_DESTROY
> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:462
> (sofia/external/03996590031055 at 203.208.207.212) State DESTROY
> 2011-06-10 11:10:30.714001 [DEBUG] mod_sofia.c:362
> sofia/external/03996590031055 at 203.208.207.212 SOFIA DESTROY
> 2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> 0x2aaab0031060 (nil)
> 2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
> 0x2aaab0031060 (nil)
> 2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> 0x2aaab00310c0 0x2aaab00b20c0
> 2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
> 0x2aaab00310c0 (nil)
> send 662 bytes to udp/[203.208.207.212]:5060 at 03:10:30.820530:
>   ------------------------------------------------------------------------
>   BYE sip:203.208.207.212:5060 SIP/2.0
>   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bK0D3Hm08XNH1Xg
>   Max-Forwards: 70
>   From: "" &lt;sip:03996563750914 at 202.73.56.46&gt;;tag=H4ZN43UH9S9Qc
>   To:
> &lt;sip:03996590031055 at 203.208.207.212
> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>   CSeq: 13501634 BYE
>   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
>   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
>   Supported: timer, precondition, path, replaces
>   Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL"
>   Content-Length: 0
>
>   ------------------------------------------------------------------------
> 2011-06-10 11:10:30.715878 [INFO] mod_com_g729.c:83 ENCODER DESTROY -
> 0x2aaab00310c0 0x2aaab00b20c0
> 2011-06-10 11:10:30.715878 [DEBUG] switch_core_state_machine.c:60
> sofia/external/03996590031055 at 203.208.207.212 Standard DESTROY
> 2011-06-10 11:10:30.715878 [DEBUG] switch_core_state_machine.c:462
> (sofia/external/03996590031055 at 203.208.207.212) State DESTROY going to
> sleep
> recv 383 bytes from udp/[203.208.207.212]:5060 at 03:10:30.823302:
>   ------------------------------------------------------------------------
>   SIP/2.0 200 Ok
>   Via: SIP/2.0/UDP
> 10.1.1.46:5080;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>   To:
> &lt;sip:03996563750914 at 203.208.207.212
> &gt;;tag=2QGB951HCR30000E1D00000u00000001QXU3LU
>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>   CSeq: 13501649 CANCEL
>   Contact: <sip:203.208.207.212:5060>
>   Content-Length:     0
>
>   ------------------------------------------------------------------------
> recv 411 bytes from udp/[203.208.207.212]:5060 at 03:10:30.824765:
>   ------------------------------------------------------------------------
>   SIP/2.0 487 Request Terminated
>   Via: SIP/2.0/UDP
> 10.1.1.46:5080;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>   To:
> &lt;sip:03996563750914 at 203.208.207.212
> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>   CSeq: 13501649 INVITE
>   Reason: SIP;cause=487;text="Request Terminated"
>   Content-Length:     0
>
>   ------------------------------------------------------------------------
> send 371 bytes to udp/[203.208.207.212]:5060 at 03:10:30.824891:
>   ------------------------------------------------------------------------
>   ACK sip:03996563750914 at 203.208.207.212 SIP/2.0
>   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKZ49rj5Qtr8aBN
>   Max-Forwards: 70
>   From: "" &lt;sip:03996590031055 at 202.73.56.46&gt;;tag=jDSe6ycN62Zar
>   To:
> &lt;sip:03996563750914 at 203.208.207.212
> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>   CSeq: 13501649 ACK
>   Content-Length: 0
>
>   ------------------------------------------------------------------------
> recv 380 bytes from udp/[203.208.207.212]:5060 at 03:10:30.826375:
>   ------------------------------------------------------------------------
>   SIP/2.0 200 Ok
>   Via: SIP/2.0/UDP
> 10.1.1.46:5080;rport=5080;branch=z9hG4bK0D3Hm08XNH1Xg;received=10.1.1.46
>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>   To:
> &lt;sip:03996590031055 at 203.208.207.212
> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>   CSeq: 13501634 BYE
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Leg-A-is-automatically-disconnected-on-Leg-B-orginate-failure-tp6482192p6482192.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Fri Jun 17 00:55:46 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 16 Jun 2011 13:55:46 -0700
Subject: [Freeswitch-users] using keypress to stop a recording.
In-Reply-To: <4DFA517C.6010109@499x.com>
References: <4DFA517C.6010109@499x.com>
Message-ID: <BANLkTimScEmhFtq4sW44CU5LPCifpcEoiQ@mail.gmail.com>

How about setting this only to #?
http://wiki.freeswitch.org/wiki/Channel_Variables#playback_terminators

-MC

On Thu, Jun 16, 2011 at 11:54 AM, Wes <wes-fs at 499x.com> wrote:

> I'm using the following script to record, and I"m hoping to take a user
> keypress to trigger the end of the recording and then play it back:
>
> (the problem is that it stops on ANY keypress, while it seems like it
> should only break on a "#" keypress)
>
> I found this example at:
> http://wiki.freeswitch.org/wiki/Mod_lua#session:recordFile
>
> local numberToCall = 1234
> local session = freeswitch.Session("sofia/xxx.xxx.xxx.xxx/"..numberToCall);
> session:set_tts_parms("flite", "kal");
> session:speak("Thank you for using my recording service, press the pound
> key to stop the recording."  );
>
> function onInput(s, type, obj)
>   if (type == "dtmf" and obj['digit'] == '#') then
>       return "break";
>   end
> end
>
> session:setInputCallback("onInput", "");
> session:recordFile("/tmp/luatest.wav");
> freeswitch.consoleLog("info", "recording stopped by user keypress \n");
> session:speak("your voicehas been recorded, i will play it for you now");
> session:streamFile("/tmp/luatest.wav");
> session:speak("that's it, goodbye.");
> freeswitch.consoleLog("info", "hanging up... \n");
> session:hangup();
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dave at clancysystems.com  Fri Jun 17 01:18:06 2011
From: dave at clancysystems.com (Dave)
Date: Thu, 16 Jun 2011 15:18:06 -0600
Subject: [Freeswitch-users] Setting Up an Sip Trunk without
	Authentication
References: <75EE58D2AC7E4BFE9A53B699375DD699@clancysystems.com><BANLkTinwXNF4zyONuvBOpiDmrXwqYiv=Yw@mail.gmail.com><5A9A76E6D31143F1B2C971E2A7508D22@clancysystems.com><BANLkTimg6hEv0Vbg_+WkjKcuADp5dz0hNA@mail.gmail.com><AF706099B82D4F309EDBEA0ACE7DA5D4@clancysystems.com>
	<BANLkTimR8-x4yWk=QFymZymVVK2JdPt_-g@mail.gmail.com>
Message-ID: <0580E32182F04516A074B86F364EDB5E@clancysystems.com>

I did both, and even restarted the service. same results. Velocity did a call capture on thier end and sent it to me. They say that FreeSWITCH is requiring proxy authentication and that that is the issue.
SIP Status: 407 Proxy Authentication Required


  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Thursday, June 16, 2011 2:46 PM
  Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication


  Did you reloadxml and restart the sofia profile after the changes? Also, you need to "reloadacl" after doing a change to acl.conf.xml.


  -MC



  On Thu, Jun 16, 2011 at 1:27 PM, Dave <dave at clancysystems.com> wrote:

    Yes, I am only testing inbound.  When I call from our office phone (not connected to FreeSWITCH, it's on a trunk from broadvox)  I Get a busy signal and at that moment FreeSWITCH shows...
    "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone at theirIP] from their IP.

    If I call from my cell phone I get the Operator Message "your call did not go through" and at that moment the same thing shows in FreeSWITCH
    "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone at theirIP] from their IP.
     

      ----- Original Message ----- 
      From: Michael Collins 
      To: FreeSWITCH Users Help 
      Sent: Thursday, June 16, 2011 2:11 PM
      Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication





      On Thu, Jun 16, 2011 at 1:02 PM, Dave <dave at clancysystems.com> wrote:


        Under "domains" in acl.conf.xml I added  

           <node type="allow" value="xxx.xxx.xxx.xxx"/>

      The correct syntax is:
      <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>

        x's are the IP address it's just a single address not a CIDR.


        And I created the file velocity_did.xml with the following code..

        <include>
          <extension name="velocity_did"> 

            <condition field="destination_number" expression="^(2064001950)$">

              <action application="set" data="domain_name=$${domain}"/>
              <action application="answer"/> 

           <action application="lua" data="cme_ivr.lua channel_name"/>
         
         </condition>
          </extension>

        </include>

        I still just get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone at theirIP] from their IP, but nothing more.

      Wait - can you explain *exactly* what you're doing for testing? Are you trying to dial out via Velocity and back in to your DID? If so then you have 2 completely different things you need to set up. The instructions I gave were only for inbound DID, so use a cell phone or something to test that.


      For outbound it seems like they are sending you an auth challenge, which means they need to give you a username and password as well as a host/ip. You need to create a gateway, preferably in your external profile. Just be sure to set the "register" param to false since they are not expecting you to register with them. 


      -MC


        Do I need to change the line ..

        <action application="set" data="domain_name=$${domain}"/>

        to the IP address? If so what syntax?

        They have no Domain Name.

        Thanks again for your help.

        Dave Goodwin      

          ----- Original Message ----- 
          From: Michael Collins 
          To: FreeSWITCH Users Help 
          Sent: Thursday, June 16, 2011 12:15 PM
          Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication


          Dave, 


          If you are simply handling incoming calls then you will need to do two things:


          Add Velocity's IP addr to the domains section of acl.conf.xml
          Add an extension to the public context (conf/dialplan/public.xml)


          Allowing a call into FS via an ACL will send it into the "public" context; from there you need to transfer it to the default context or just send it straight to your Lua script.


          Also, I don't believe you need to use a "bridge" app based on the description you gave. Bridge is used to create a new outbound call leg (B leg) and connect it to the inbound call leg (A leg). If you just are handling a call with an IVR then there is no B leg needed, thus no need to bridge.


          Welcome to FreeSWITCH, btw! :) I can highly recommend the FS book (the "bridge" book, ironically) as it discusses some of these basic concepts that will make your FS experience a whole lot more pleasant.


          -MC


          On Thu, Jun 16, 2011 at 8:54 AM, Dave <dave at clancysystems.com> wrote:

            Hi,

            I am new to FreeSWITCH and need to set up FreeSWITCH to receive calls through a Velocity Networks SIP trunk. They say I should not authenticate with Username and Password. Rather, I connect directly to the their IP address.

            I put

            <extension name="velocity">
                  <condition field="destination_number" expression="^2064001950$">
                    <action application="bridge" data="sofia/external/$ {destination_number}@their_ip_address"/>
                    <action application="lua" data="cme_ivr.lua channel_name"/>
                 </condition>
            </extension>

            in the default.xml under Dialplan

            When I Dial the DID from a Phone (not one connected to the FreeSWITCH server) I get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone at theirIP] from their IP, but nothing more.

            What I hope to do is take the incoming call from 2064001950 and route it to the Lua IVR script above.

            I Appreciate any help you may offer.

            Dave Goodwin





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





        _______________________________________________
        Join us at ClueCon 2011, Aug 9-11, Chicago
        http://www.cluecon.com 877-7-4ACLUE
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org





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

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



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From msc at freeswitch.org  Fri Jun 17 01:22:50 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 16 Jun 2011 14:22:50 -0700
Subject: [Freeswitch-users] Setting Up an Sip Trunk without
	Authentication
In-Reply-To: <0580E32182F04516A074B86F364EDB5E@clancysystems.com>
References: <75EE58D2AC7E4BFE9A53B699375DD699@clancysystems.com>
	<BANLkTinwXNF4zyONuvBOpiDmrXwqYiv=Yw@mail.gmail.com>
	<5A9A76E6D31143F1B2C971E2A7508D22@clancysystems.com>
	<BANLkTimg6hEv0Vbg_+WkjKcuADp5dz0hNA@mail.gmail.com>
	<AF706099B82D4F309EDBEA0ACE7DA5D4@clancysystems.com>
	<BANLkTimR8-x4yWk=QFymZymVVK2JdPt_-g@mail.gmail.com>
	<0580E32182F04516A074B86F364EDB5E@clancysystems.com>
Message-ID: <BANLkTi=+aniSP1qUpVejYc-hksk51V8wMA@mail.gmail.com>

Okay, I want you to go read up on gather data and dropping it on pastebin.
The skills you hone there will server you well. :)

http://wiki.freeswitch.org/wiki/Reporting_Bugs

Need a debug log and a sip trace. Drop it on pastebin. Hint: use "sofia
global siptrace on" to get all sip traffic logged to the console.

Put the pb link in this email thread.
-MC

On Thu, Jun 16, 2011 at 2:18 PM, Dave <dave at clancysystems.com> wrote:

>  I did both, and even restarted the service. same results. Velocity did a
> call capture on thier end and sent it to me. They say that FreeSWITCH is
> requiring proxy authentication and that that is the issue.
> SIP Status: 407 Proxy Authentication Required
>
>
>
> ----- Original Message -----
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Thursday, June 16, 2011 2:46 PM
> *Subject:* Re: [Freeswitch-users] Setting Up an Sip Trunk without
> Authentication
>
> Did you reloadxml and restart the sofia profile after the changes? Also,
> you need to "reloadacl" after doing a change to acl.conf.xml.
>
> -MC
>
> On Thu, Jun 16, 2011 at 1:27 PM, Dave <dave at clancysystems.com> wrote:
>
>>  Yes, I am only testing inbound.  When I call from our office phone (not
>> connected to FreeSWITCH, it's on a trunk from broadvox)  I Get a busy signal
>> and at that moment FreeSWITCH shows...
>>  "SIP auth challenge (INVITE) on sofia profile 'internal' for
>> [myphone at theirIP] from their IP.
>>
>> If I call from my cell phone I get the Operator Message "your call did not
>> go through" and at that moment the same thing shows in FreeSWITCH
>>  "SIP auth challenge (INVITE) on sofia profile 'internal' for
>> [myphone at theirIP] from their IP.
>>
>>
>>
>>  ----- Original Message -----
>> *From:* Michael Collins <msc at freeswitch.org>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>   *Sent:* Thursday, June 16, 2011 2:11 PM
>> *Subject:* Re: [Freeswitch-users] Setting Up an Sip Trunk without
>> Authentication
>>
>>
>>
>> On Thu, Jun 16, 2011 at 1:02 PM, Dave <dave at clancysystems.com> wrote:
>>
>>>
>>> Under "domains" in acl.conf.xml I added
>>>
>>>    <node type="allow" value="xxx.xxx.xxx.xxx"/>
>>>
>>>
>> The correct syntax is:
>> <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>
>>
>>
>>>  x's are the IP address it's just a single address not a CIDR.
>>>
>>>
>>> And I created the file velocity_did.xml with the following code..
>>>
>>> <include>
>>>   <extension name="velocity_did">
>>>
>>>     <condition field="destination_number" expression="^(2064001950)$">
>>>       <action application="set" data="domain_name=$${domain}"/>
>>>       <action application="answer"/>
>>>
>>>    <action application="lua" data="cme_ivr.lua channel_name"/>
>>>
>>>  </condition>
>>>   </extension>
>>> </include>
>>> I still just get the "SIP auth challenge (INVITE) on sofia profile
>>> 'internal' for [myphone at theirIP] from their IP, but nothing more.
>>>
>>>
>> Wait - can you explain *exactly* what you're doing for testing? Are you
>> trying to dial out via Velocity and back in to your DID? If so then you have
>> 2 completely different things you need to set up. The instructions I gave
>> were only for inbound DID, so use a cell phone or something to test that.
>>
>> For outbound it seems like they are sending you an auth challenge, which
>> means they need to give you a username and password as well as a host/ip.
>> You need to create a gateway, preferably in your external profile. Just be
>> sure to set the "register" param to false since they are not expecting you
>> to register with them.
>>
>> -MC
>>
>>  Do I need to change the line ..
>>>
>>> <action application="set" data="domain_name=$${domain}"/>
>>> to the IP address? If so what syntax?
>>>
>>> They have no Domain Name.
>>>
>>> Thanks again for your help.
>>>
>>> Dave Goodwin
>>>
>>>
>>> ----- Original Message -----
>>> *From:* Michael Collins <msc at freeswitch.org>
>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> *Sent:* Thursday, June 16, 2011 12:15 PM
>>> *Subject:* Re: [Freeswitch-users] Setting Up an Sip Trunk without
>>> Authentication
>>>
>>> Dave,
>>>
>>> If you are simply handling incoming calls then you will need to do two
>>> things:
>>>
>>> Add Velocity's IP addr to the domains section of acl.conf.xml
>>> Add an extension to the public context (conf/dialplan/public.xml)
>>>
>>> Allowing a call into FS via an ACL will send it into the "public"
>>> context; from there you need to transfer it to the default context or just
>>> send it straight to your Lua script.
>>>
>>> Also, I don't believe you need to use a "bridge" app based on the
>>> description you gave. Bridge is used to create a new outbound call leg (B
>>> leg) and connect it to the inbound call leg (A leg). If you just are
>>> handling a call with an IVR then there is no B leg needed, thus no need to
>>> bridge.
>>>
>>> Welcome to FreeSWITCH, btw! :) I can highly recommend the FS book (the
>>> "bridge" book, ironically) as it discusses some of these basic concepts that
>>> will make your FS experience a whole lot more pleasant.
>>>
>>> -MC
>>>
>>> On Thu, Jun 16, 2011 at 8:54 AM, Dave <dave at clancysystems.com> wrote:
>>>
>>>>  Hi,
>>>>
>>>> I am new to FreeSWITCH and need to set up FreeSWITCH to receive calls
>>>> through a Velocity Networks SIP trunk. They say I should not authenticate
>>>> with Username and Password. Rather, I connect directly to the their IP
>>>> address.
>>>>
>>>> I put
>>>>
>>>> <extension name="velocity">
>>>>       <condition field="destination_number" expression="^2064001950$">
>>>>         <action application="bridge" data="sofia/external/$
>>>> {destination_number}@their_ip_address"/>
>>>>         <action application="lua" data="cme_ivr.lua channel_name"/>
>>>>      </condition>
>>>> </extension>
>>>>
>>>> in the default.xml under Dialplan
>>>>
>>>> When I Dial the DID from a Phone (not one connected to the FreeSWITCH
>>>> server) I get the "SIP auth challenge (INVITE) on sofia profile 'internal'
>>>> for [myphone at theirIP] from their IP, but nothing more.
>>>>
>>>> What I hope to do is take the incoming call from 2064001950 and route it
>>>> to the Lua IVR script above.
>>>>
>>>> I Appreciate any help you may offer.
>>>>
>>>> Dave Goodwin
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From wes-fs at 499x.com  Fri Jun 17 01:29:37 2011
From: wes-fs at 499x.com (Wes)
Date: Thu, 16 Jun 2011 16:29:37 -0500
Subject: [Freeswitch-users] using keypress to stop a recording.
In-Reply-To: <BANLkTimScEmhFtq4sW44CU5LPCifpcEoiQ@mail.gmail.com>
References: <4DFA517C.6010109@499x.com>
	<BANLkTimScEmhFtq4sW44CU5LPCifpcEoiQ@mail.gmail.com>
Message-ID: <4DFA75C1.3050008@499x.com>

I'm writing a lua script, that link you mention doesn't seem like 
something I can do in LUA...

the problem is in this function:

function onInput(s, type, obj)
   if (type == "dtmf" and obj['digit'] == '#') then
       return "break";
   end
end

which seems to break on any keypress, so the check for # doesn't work as 
expected.

is there a different way to do this with a lua script?  Ideally, I'd 
like to do more than just break, I'd like the user to be able to review 
the message more than once, and then finally hit another key to submit 
it for real.  Like a voicemail system, actually.

On 6/16/2011 3:55 PM, Michael Collins wrote:
> How about setting this only to #?
> http://wiki.freeswitch.org/wiki/Channel_Variables#playback_terminators
>
> -MC
>
> On Thu, Jun 16, 2011 at 11:54 AM, Wes <wes-fs at 499x.com 
> <mailto:wes-fs at 499x.com>> wrote:
>
>     I'm using the following script to record, and I"m hoping to take a
>     user
>     keypress to trigger the end of the recording and then play it back:
>
>     (the problem is that it stops on ANY keypress, while it seems like it
>     should only break on a "#" keypress)
>
>     I found this example at:
>     http://wiki.freeswitch.org/wiki/Mod_lua#session:recordFile
>
>     local numberToCall = 1234
>     local session =
>     freeswitch.Session("sofia/xxx.xxx.xxx.xxx/"..numberToCall);
>     session:set_tts_parms("flite", "kal");
>     session:speak("Thank you for using my recording service, press the
>     pound
>     key to stop the recording."  );
>
>     function onInput(s, type, obj)
>       if (type == "dtmf" and obj['digit'] == '#') then
>           return "break";
>       end
>     end
>
>     session:setInputCallback("onInput", "");
>     session:recordFile("/tmp/luatest.wav");
>     freeswitch.consoleLog("info", "recording stopped by user keypress
>     \n");
>     session:speak("your voicehas been recorded, i will play it for you
>     now");
>     session:streamFile("/tmp/luatest.wav");
>     session:speak("that's it, goodbye.");
>     freeswitch.consoleLog("info", "hanging up... \n");
>     session:hangup();
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From msc at freeswitch.org  Fri Jun 17 01:32:24 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 16 Jun 2011 14:32:24 -0700
Subject: [Freeswitch-users] using keypress to stop a recording.
In-Reply-To: <4DFA75C1.3050008@499x.com>
References: <4DFA517C.6010109@499x.com>
	<BANLkTimScEmhFtq4sW44CU5LPCifpcEoiQ@mail.gmail.com>
	<4DFA75C1.3050008@499x.com>
Message-ID: <BANLkTikY45-Pys5+9rY48nT_+Zx+Vxfokg@mail.gmail.com>

It's just a channel variable. Set it prior to calling the Lua script or use
session:setVariable("playback_terminators","#")

-MC

On Thu, Jun 16, 2011 at 2:29 PM, Wes <wes-fs at 499x.com> wrote:

>  I'm writing a lua script, that link you mention doesn't seem like
> something I can do in LUA...
>
> the problem is in this function:
>
>
> function onInput(s, type, obj)
>   if (type == "dtmf" and obj['digit'] == '#') then
>       return "break";
>   end
> end
>
> which seems to break on any keypress, so the check for # doesn't work as
> expected.
>
> is there a different way to do this with a lua script?  Ideally, I'd like
> to do more than just break, I'd like the user to be able to review the
> message more than once, and then finally hit another key to submit it for
> real.  Like a voicemail system, actually.
>
>
> On 6/16/2011 3:55 PM, Michael Collins wrote:
>
> How about setting this only to #?
> http://wiki.freeswitch.org/wiki/Channel_Variables#playback_terminators
>
>  -MC
>
> On Thu, Jun 16, 2011 at 11:54 AM, Wes <wes-fs at 499x.com> wrote:
>
>> I'm using the following script to record, and I"m hoping to take a user
>> keypress to trigger the end of the recording and then play it back:
>>
>> (the problem is that it stops on ANY keypress, while it seems like it
>> should only break on a "#" keypress)
>>
>> I found this example at:
>> http://wiki.freeswitch.org/wiki/Mod_lua#session:recordFile
>>
>> local numberToCall = 1234
>> local session =
>> freeswitch.Session("sofia/xxx.xxx.xxx.xxx/"..numberToCall);
>> session:set_tts_parms("flite", "kal");
>> session:speak("Thank you for using my recording service, press the pound
>> key to stop the recording."  );
>>
>> function onInput(s, type, obj)
>>   if (type == "dtmf" and obj['digit'] == '#') then
>>       return "break";
>>   end
>> end
>>
>> session:setInputCallback("onInput", "");
>> session:recordFile("/tmp/luatest.wav");
>> freeswitch.consoleLog("info", "recording stopped by user keypress \n");
>> session:speak("your voicehas been recorded, i will play it for you now");
>> session:streamFile("/tmp/luatest.wav");
>> session:speak("that's it, goodbye.");
>> freeswitch.consoleLog("info", "hanging up... \n");
>> session:hangup();
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From justlikeef at gmail.com  Fri Jun 17 01:40:17 2011
From: justlikeef at gmail.com (Rob Hutton)
Date: Thu, 16 Jun 2011 17:40:17 -0400
Subject: [Freeswitch-users] SRTP
In-Reply-To: <BANLkTikchSX+zGw7MfYiLDicdLJ4jo6O+A@mail.gmail.com>
References: <201106152210.41334.justlikeef@gmail.com>
	<201106161322.53638.justlikeef@gmail.com>
	<BANLkTikchSX+zGw7MfYiLDicdLJ4jo6O+A@mail.gmail.com>
Message-ID: <201106161740.18059.justlikeef@gmail.com>

I am trying to get encryption working from within Bluebox, in the most  
"reasonably flexible" way possible. (So no, not the default dialpan, but I 
missed the example so I will go back and look at it)

So, one scenario I am thinking needs to be supported is where you have two 
devices that are registered to the same user, one encrypted and one not.  For 
instance, a phone and a remote ringer.

What I am looking for is the best way to stay as flexible as possible.  It may 
be a situation where you end up turning on encryption system wide if the 
devices support it, but that is overkill in a situation where there is a 
seperate voice and data VLAN unless there is a need for that level of 
security..

It may be a situation where I need to offer both options and write two 
dialplan enries in the situation where the admin wants to enable it device by 
device.

BTW, I am also using my head to beat through getting TLS working on the front 
end.  I would REALLY appreciate another set of eyes if you have time.  

http://jira.freeswitch.org/browse/FS-3346?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-
tabpanel&focusedCommentId=24719#action_24719

Thanks,
Rob

On Thursday 16 June 2011 14:01:41 Michael Collins wrote:
> Are you working off of the default.xml dialplan file? If so, it has an
> example condition already:
> 
>       <condition field="${sip_has_crypto}"
> expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$"
> break="never">
> <action application="set" data="sip_secure_media=true"/>
> <!-- Offer SRTP on outbound legs if we have it on inbound. -->
> <!-- <action application="export" data="sip_secure_media=true"/> -->
>       </condition>
> 
> What exactly are you checking on in your scenario? Most likely there is an
> elegant way to do it. Give us the plain language description of the problem
> you're addressing and the community will no doubt have good suggestions for
> you.
> 
> -MC
> 
> On Thu, Jun 16, 2011 at 10:22 AM, Rob Hutton <justlikeef at gmail.com> wrote:
> > Steven -
> > 
> > Thanks for the help here...
> > 
> > So there would have to be two dialplan entries for this number to work
> > with either RTP or SRTP? (Maybe two devices registering to the same
> > user?)
> > 
> > Would it make more since to do this in a more global manner higher up in
> > the
> > dialplan in its own condition block?
> > 
> > On Thursday 16 June 2011 03:15:33 Steven Ayre wrote:
> > >  {ERR} mod_dialplan_xml.c:110 Nexted conditions are not allowed
> > > 
> > > That's because it shouldn't be nested. It's not missing a /, and the
> > > 1st Should have the /. The extra indendation shouldn't be there on the
> > > 2nd.
> > > 
> > > It should look like this:
> > > 
> > > <extension name="incoming-fxs">
> > > 
> > >      <condition field="destination_number" expression="^(202)$"/>
> > >      <condition field="${sip_has_crypto}"
> > > 
> > > expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$"
> > > break="never">
> > > 
> > >           <action application="set" data="sip_secure_media=true"/>
> > >           <action application="bridge" data="openzap/1/1"/>
> > >      
> > >      </condition>
> > > 
> > > </extension>
> > > 
> > > The two conditions function as an AND, even though it's not nested. FS
> > > stops checking the extension as soon as it sees a condition that's
> > > false (at least by default and in the above case), so if the
> > > destination is not 202 it'll never get to the 2nd condition.
> > > 
> > > -Steve
> > > 
> > > On 16 June 2011 03:10, Rob Hutton <justlikeef at gmail.com> wrote:
> > > > I think I have TLS and SRTP working at this point, but in the docs it
> > > > says to use the following template for the dialplan:
> > > > 
> > > > http://wiki.freeswitch.org/wiki/Secure_RTP:
> > > >  <extension name="incoming-fxs">
> > > >  
> > > >    <condition field="destination_number" expression="^(202)$"/>
> > > >    
> > > >      <condition field="${sip_has_crypto}"
> > > > 
> > > > expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$"
> > > > break="never">
> > > > 
> > > >      <action application="set" data="sip_secure_media=true"/>
> > > >      <action application="bridge" data="openzap/1/1"/>
> > > >    
> > > >    </condition>
> > > >  
> > > >  </extension
> > > > 
> > > > 1) There is a missing > at the end of the close extension tag.
> > > > 2) There is either a missing / at the end of the internal condition
> > 
> > line,
> > 
> > > > or a missing condition close tag somewhere
> > > > 3) When I fix the interal condition, I get an error:
> > > > 
> > > > {ERR} mod_dialplan_xml.c:110 Nexted conditions are not allowed
> > > > 
> > > > All this, but a packet capture shows that SRTP is working based on
> > > > what
> > 
> > I
> > 
> > > > did on:
> > > > 
> > > > http://wiki.freeswitch.org/wiki/SIP_TLS
> > > > 
> > > > Can someone give me some guidance on  the Secure_RTP page and I will
> > > > update whatever?
> > > > 
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > 
> > > > UNSUBSCRIBE:
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 
> > > > http://www.freeswitch.org
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org


From dave at clancysystems.com  Fri Jun 17 01:46:23 2011
From: dave at clancysystems.com (Dave)
Date: Thu, 16 Jun 2011 15:46:23 -0600
Subject: [Freeswitch-users] Setting Up an Sip Trunk without
	Authentication
References: <75EE58D2AC7E4BFE9A53B699375DD699@clancysystems.com><BANLkTinwXNF4zyONuvBOpiDmrXwqYiv=Yw@mail.gmail.com><5A9A76E6D31143F1B2C971E2A7508D22@clancysystems.com><BANLkTimg6hEv0Vbg_+WkjKcuADp5dz0hNA@mail.gmail.com><AF706099B82D4F309EDBEA0ACE7DA5D4@clancysystems.com><BANLkTimR8-x4yWk=QFymZymVVK2JdPt_-g@mail.gmail.com><0580E32182F04516A074B86F364EDB5E@clancysystems.com>
	<BANLkTi=+aniSP1qUpVejYc-hksk51V8wMA@mail.gmail.com>
Message-ID: <271B8C7B3AE54D7CA9229E49D3AB652D@clancysystems.com>

Will Do. Thanks very much for your help.  I bought the book:)

Dave

  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Thursday, June 16, 2011 3:22 PM
  Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication


  Okay, I want you to go read up on gather data and dropping it on pastebin. The skills you hone there will server you well. :)


  http://wiki.freeswitch.org/wiki/Reporting_Bugs


  Need a debug log and a sip trace. Drop it on pastebin. Hint: use "sofia global siptrace on" to get all sip traffic logged to the console.


  Put the pb link in this email thread.
  -MC


  On Thu, Jun 16, 2011 at 2:18 PM, Dave <dave at clancysystems.com> wrote:

    I did both, and even restarted the service. same results. Velocity did a call capture on thier end and sent it to me. They say that FreeSWITCH is requiring proxy authentication and that that is the issue.
    SIP Status: 407 Proxy Authentication Required 


      ----- Original Message ----- 
      From: Michael Collins 
      To: FreeSWITCH Users Help 
      Sent: Thursday, June 16, 2011 2:46 PM
      Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication


      Did you reloadxml and restart the sofia profile after the changes? Also, you need to "reloadacl" after doing a change to acl.conf.xml. 


      -MC



      On Thu, Jun 16, 2011 at 1:27 PM, Dave <dave at clancysystems.com> wrote:

        Yes, I am only testing inbound.  When I call from our office phone (not connected to FreeSWITCH, it's on a trunk from broadvox)  I Get a busy signal and at that moment FreeSWITCH shows...
        "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone at theirIP] from their IP.

        If I call from my cell phone I get the Operator Message "your call did not go through" and at that moment the same thing shows in FreeSWITCH
        "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone at theirIP] from their IP.
         

          ----- Original Message ----- 
          From: Michael Collins 
          To: FreeSWITCH Users Help 
          Sent: Thursday, June 16, 2011 2:11 PM
          Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication





          On Thu, Jun 16, 2011 at 1:02 PM, Dave <dave at clancysystems.com> wrote:


            Under "domains" in acl.conf.xml I added  

               <node type="allow" value="xxx.xxx.xxx.xxx"/>

          The correct syntax is:
          <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>

            x's are the IP address it's just a single address not a CIDR.


            And I created the file velocity_did.xml with the following code..

            <include>
              <extension name="velocity_did"> 

                <condition field="destination_number" expression="^(2064001950)$">

                  <action application="set" data="domain_name=$${domain}"/>
                  <action application="answer"/> 

               <action application="lua" data="cme_ivr.lua channel_name"/>
             
             </condition>
              </extension>

            </include>

            I still just get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone at theirIP] from their IP, but nothing more.

          Wait - can you explain *exactly* what you're doing for testing? Are you trying to dial out via Velocity and back in to your DID? If so then you have 2 completely different things you need to set up. The instructions I gave were only for inbound DID, so use a cell phone or something to test that.


          For outbound it seems like they are sending you an auth challenge, which means they need to give you a username and password as well as a host/ip. You need to create a gateway, preferably in your external profile. Just be sure to set the "register" param to false since they are not expecting you to register with them. 


          -MC


            Do I need to change the line ..

            <action application="set" data="domain_name=$${domain}"/>

            to the IP address? If so what syntax?

            They have no Domain Name.

            Thanks again for your help.

            Dave Goodwin      

              ----- Original Message ----- 
              From: Michael Collins 
              To: FreeSWITCH Users Help 
              Sent: Thursday, June 16, 2011 12:15 PM
              Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication


              Dave, 


              If you are simply handling incoming calls then you will need to do two things:


              Add Velocity's IP addr to the domains section of acl.conf.xml
              Add an extension to the public context (conf/dialplan/public.xml)


              Allowing a call into FS via an ACL will send it into the "public" context; from there you need to transfer it to the default context or just send it straight to your Lua script.


              Also, I don't believe you need to use a "bridge" app based on the description you gave. Bridge is used to create a new outbound call leg (B leg) and connect it to the inbound call leg (A leg). If you just are handling a call with an IVR then there is no B leg needed, thus no need to bridge.


              Welcome to FreeSWITCH, btw! :) I can highly recommend the FS book (the "bridge" book, ironically) as it discusses some of these basic concepts that will make your FS experience a whole lot more pleasant.


              -MC


              On Thu, Jun 16, 2011 at 8:54 AM, Dave <dave at clancysystems.com> wrote:

                Hi,

                I am new to FreeSWITCH and need to set up FreeSWITCH to receive calls through a Velocity Networks SIP trunk. They say I should not authenticate with Username and Password. Rather, I connect directly to the their IP address.

                I put

                <extension name="velocity">
                      <condition field="destination_number" expression="^2064001950$">
                        <action application="bridge" data="sofia/external/$ {destination_number}@their_ip_address"/>
                        <action application="lua" data="cme_ivr.lua channel_name"/>
                     </condition>
                </extension>

                in the default.xml under Dialplan

                When I Dial the DID from a Phone (not one connected to the FreeSWITCH server) I get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone at theirIP] from their IP, but nothing more.

                What I hope to do is take the incoming call from 2064001950 and route it to the Lua IVR script above.

                I Appreciate any help you may offer.

                Dave Goodwin





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





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

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        FreeSWITCH-users at lists.freeswitch.org
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    _______________________________________________
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    http://www.cluecon.com 877-7-4ACLUE

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From wes-fs at 499x.com  Fri Jun 17 01:52:45 2011
From: wes-fs at 499x.com (Wes)
Date: Thu, 16 Jun 2011 16:52:45 -0500
Subject: [Freeswitch-users] using keypress to stop a recording.
In-Reply-To: <BANLkTikY45-Pys5+9rY48nT_+Zx+Vxfokg@mail.gmail.com>
References: <4DFA517C.6010109@499x.com>	<BANLkTimScEmhFtq4sW44CU5LPCifpcEoiQ@mail.gmail.com>	<4DFA75C1.3050008@499x.com>
	<BANLkTikY45-Pys5+9rY48nT_+Zx+Vxfokg@mail.gmail.com>
Message-ID: <4DFA7B2D.5040005@499x.com>

ok, tried that, but it didn't change the behavior  In fact, I removed 
both the onInput function, and the call to session:setVariable, and the 
recording is still halted by *any* keypress on the keypad.  So this must 
be the default behavior of how to stop the recording...

Can anyone confirm this?

Thanks.


On 6/16/2011 4:32 PM, Michael Collins wrote:
> It's just a channel variable. Set it prior to calling the Lua script 
> or use session:setVariable("playback_terminators","#")
>
> -MC
>
> On Thu, Jun 16, 2011 at 2:29 PM, Wes <wes-fs at 499x.com 
> <mailto:wes-fs at 499x.com>> wrote:
>
>     I'm writing a lua script, that link you mention doesn't seem like
>     something I can do in LUA...
>
>     the problem is in this function:
>
>
>     function onInput(s, type, obj)
>       if (type == "dtmf" and obj['digit'] == '#') then
>           return "break";
>       end
>     end
>
>     which seems to break on any keypress, so the check for # doesn't
>     work as expected.
>
>     is there a different way to do this with a lua script?  Ideally,
>     I'd like to do more than just break, I'd like the user to be able
>     to review the message more than once, and then finally hit another
>     key to submit it for real.  Like a voicemail system, actually.
>
>
>     On 6/16/2011 3:55 PM, Michael Collins wrote:
>>     How about setting this only to #?
>>     http://wiki.freeswitch.org/wiki/Channel_Variables#playback_terminators
>>
>>     -MC
>>
>>     On Thu, Jun 16, 2011 at 11:54 AM, Wes <wes-fs at 499x.com
>>     <mailto:wes-fs at 499x.com>> wrote:
>>
>>         I'm using the following script to record, and I"m hoping to
>>         take a user
>>         keypress to trigger the end of the recording and then play it
>>         back:
>>
>>         (the problem is that it stops on ANY keypress, while it seems
>>         like it
>>         should only break on a "#" keypress)
>>
>>         I found this example at:
>>         http://wiki.freeswitch.org/wiki/Mod_lua#session:recordFile
>>
>>         local numberToCall = 1234
>>         local session =
>>         freeswitch.Session("sofia/xxx.xxx.xxx.xxx/"..numberToCall);
>>         session:set_tts_parms("flite", "kal");
>>         session:speak("Thank you for using my recording service,
>>         press the pound
>>         key to stop the recording."  );
>>
>>         function onInput(s, type, obj)
>>           if (type == "dtmf" and obj['digit'] == '#') then
>>               return "break";
>>           end
>>         end
>>
>>         session:setInputCallback("onInput", "");
>>         session:recordFile("/tmp/luatest.wav");
>>         freeswitch.consoleLog("info", "recording stopped by user
>>         keypress \n");
>>         session:speak("your voicehas been recorded, i will play it
>>         for you now");
>>         session:streamFile("/tmp/luatest.wav");
>>         session:speak("that's it, goodbye.");
>>         freeswitch.consoleLog("info", "hanging up... \n");
>>         session:hangup();
>>
>>
>>         _______________________________________________
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>     _______________________________________________
>>     Join us at ClueCon 2011, Aug 9-11, Chicago
>>     http://www.cluecon.com  877-7-4ACLUE
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>     _______________________________________________
>     Join us at ClueCon 2011, Aug 9-11, Chicago
>     http://www.cluecon.com 877-7-4ACLUE
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
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From sid.kshatriya at gmail.com  Fri Jun 17 01:56:44 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Fri, 17 Jun 2011 03:26:44 +0530
Subject: [Freeswitch-users] Want to play music on hold while executing a
 time consuming operation in Lua
In-Reply-To: <BANLkTikVkBbOJLHQSzuQ4FeqphJnFTaSfw@mail.gmail.com>
References: <BANLkTin5-d2QSCGCaaOW6j+Y_xu3AfNfog@mail.gmail.com>
	<BANLkTikVkBbOJLHQSzuQ4FeqphJnFTaSfw@mail.gmail.com>
Message-ID: <BANLkTin2gS3DZWV7Y_DHYB514kEq1drHJA@mail.gmail.com>

Thanks Michael. This is quite a mainstream requirement :-) ! Not everyone
uses ESL. Is there no way we can "hack" this in Lua?

Will await your research,

Sidharth

On Thu, Jun 16, 2011 at 11:51 PM, Michael Collins <msc at freeswitch.org>wrote:

> Actually, I am looking into ways to accomplish these. They are somewhat
> related to some recipes I am doing for the FS Cookbook. Please stand by.
> FYI, I am looking at an ESL-based solution, not a dialplan script. Thus far
> I have not found a viable way to do async stuff like this from a dp script
> and I'm thinking ESL is the way to go. I will report back in a bit.
>
> -MC
>
> On Wed, Jun 15, 2011 at 9:53 PM, Sidharth Kshatriya <
> sid.kshatriya at gmail.com> wrote:
>
>> I am implementing an IVR using Lua in Freeswitch. In my Lua script I use
>> curl to a web service. Sometimes the response takes a long time to come
>> back. During that time period I would like to play music on hold.
>>
>> I have searched the freeswitch discussion archives a lot. There seem to be
>> many suggested ways to implement music on hold from a Lua script but the
>> answers are not very clear / not really applicable to my use case. I don't
>> know what method I should use and which one is recommended.
>>
>> *Method 1:* Transfer the call to 9664 (music on hold extension). However
>> the implementation for this solution for this does not seem to be available
>> in Lua. For example: How would I transfer the call back?
>> *Method 2:* Using bgapi uuid_park park the call and using uuid_broadcast
>> play an audio file. Again what do I do to unpark the call..?
>>
>> Thanks,
>>
>> Sidharth
>> [P.S. This is a repost, I apologize but I never got any responses on my
>> previous email. Need help! :-) ]
>>
>> --
>> Sidharth Kshatriya
>> www.sidk.info
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sidharth Kshatriya
www.sidk.info
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From gcd at i.ph  Fri Jun 17 02:38:30 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Fri, 17 Jun 2011 06:38:30 +0800
Subject: [Freeswitch-users] Making a external call to Mobile number
In-Reply-To: <BANLkTinFuhpjGkLv14UC3LGUYcFc8H9cjQ@mail.gmail.com>
References: <BANLkTinFuhpjGkLv14UC3LGUYcFc8H9cjQ@mail.gmail.com>
Message-ID: <BANLkTinaMATA-tCcTDi367JFK9a9YG4OAw@mail.gmail.com>

in fs_cli, execute *sofia status* to check if the account is REGED. the
account should be in  sip_profiles/external/. show us a sample of the
account entry and the dialplan.


On Fri, Jun 17, 2011 at 1:04 AM, ankIT WALiA <ankitwalia4u at gmail.com> wrote:

> Dear all,
>
> I am trying to make an external call to a mobile number of India using the
> default ext number 1005. For doing this, I did the following things.
>
> First, I created a user at IPTEL.org, I added a SIP profile. I registered
> with IPTEL. I also added a dial plan regex for 91 - (10 digit number). But,
> because of some issue, I was not able to connect to IPTEL SIP. I was getting
> error on X-LITE about some server issue.
>
> then, I tried my VOIP account with ActionVoip which is a paid account and
> has SIP service. I could connect to my action voip account through X-LITE
> SIP.
>
> I made a SIP profile for Action Voip and registered the username with
> action voip in FS. On reload, it got registered using action voip SIP
> profile. I changed the dialplan to use actionvoip gateway.
>
> But still, when I am trying to call to my mobile number in India. The call
> is not able to connect.
>
> I think there is gap in my knowledge. Please help me I am new to
> FreeSwitch.
>
> Thanks
> Ankit
>
> --
> Life is like a rose its upto u feel it as its fragrance or thorns
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rentmycoder at gmail.com  Fri Jun 17 02:56:59 2011
From: rentmycoder at gmail.com (rentmycoder rentmycoder)
Date: Fri, 17 Jun 2011 00:56:59 +0200
Subject: [Freeswitch-users] how to show unregistered users
Message-ID: <BANLkTin0AZjiJ=FBhcrmjrW5d3BdRCYqQg@mail.gmail.com>

Hi,

It may be an easy question, but I'm unable to find the answer on wiki
or google...

sofia status profie internal show only registered internal extensions...

how to list all users (both reged and unreged)?

like: sip show peers in Asterisk...

Thanks,,,


From gcd at i.ph  Fri Jun 17 03:03:50 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Fri, 17 Jun 2011 07:03:50 +0800
Subject: [Freeswitch-users] VoIP IP DSLAMs
Message-ID: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>

hello everybody,

i'm looking for small (24/48 ports)  IP DSLAMs that inter-connects directly
to FreeSwitch via IP.  is this already available on the market? or do we
still have to connect FreeSwitch via POTS splitters and FXS gateways? i
appreciate if you can mention some brands/models.

tks,
nandy
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From sascha.daniels at amooma.de  Fri Jun 17 03:59:32 2011
From: sascha.daniels at amooma.de (Sascha Daniels)
Date: Fri, 17 Jun 2011 01:59:32 +0200
Subject: [Freeswitch-users] Hooks for own call logs
Message-ID: <BANLkTinz0cptgEt3JQ8Eq2FDuNbgsfDdLw@mail.gmail.com>

Hi together.

My dialplan is generated by a java script.

I am writing call logs (not cdr) in the application that feeds the js.

Everething is fine except canceled calls.

I think I need a hook in the dialplan.

How can I make a system call (or what ever) when a call is canceled?

Any hints?

Regards Sascga
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From justlikeef at gmail.com  Fri Jun 17 05:31:04 2011
From: justlikeef at gmail.com (Rob Hutton)
Date: Thu, 16 Jun 2011 21:31:04 -0400
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
References: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
Message-ID: <201106162131.04859.justlikeef@gmail.com>

Can you explain further what you are trying to do?  A DSLAM is a device that provides DSL transport, which is independent of what you run across it.  So normally, you have

|Server | ----|network|----|DSLAM| --------[dsl line] -------[modem or modem/router combo]-----[network]---[Client]

What the Server is (Could be Freeswitch) and what the Client is (could be a VOIP phone) are independent of the transport???  Onewire, Cisco, and others make some DSL Modem/Router combos with integrated Voip to FXS ports for the CPE end.

If you find a DSLAM running one of the OSs that Freeswitch supports, you could compile Freeswitch for it, but I haven't seen one with enough CPU tp handle much besides the traffic.

On Thursday 16 June 2011 19:03:50 Nandy Dagondon wrote:
> hello everybody,
> 
> i'm looking for small (24/48 ports)  IP DSLAMs that inter-connects directly
> to FreeSwitch via IP.  is this already available on the market? or do we
> still have to connect FreeSwitch via POTS splitters and FXS gateways? i
> appreciate if you can mention some brands/models.
> 
> tks,
> nandy


From justlikeef at gmail.com  Fri Jun 17 05:32:46 2011
From: justlikeef at gmail.com (Rob Hutton)
Date: Thu, 16 Jun 2011 21:32:46 -0400
Subject: [Freeswitch-users] mod_fifo shared lag
In-Reply-To: <4DF9E16C.4090608@amooma.de>
References: <4DF9E16C.4090608@amooma.de>
Message-ID: <201106162132.47164.justlikeef@gmail.com>

The callcenter app has a "recovery" time for each queue you are a member of, 
but I'm not sure about fifo.

On Thursday 16 June 2011 06:56:44 Sascha Daniels wrote:
> Hi together.
> 
> Is there a possibility to have a shared lag over several fifos?
> 
> In a callcenter with agents in several Queues the lag is 10sec. After a
> call from fifo A the agent should not get a call from fifo B within
> those 10sec.
> 
> Basically like shared_lastcall=yes in Asterisk 1.6
> 
> Regards
> 
>                     Sascha


From msc at freeswitch.org  Fri Jun 17 05:43:21 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 16 Jun 2011 18:43:21 -0700
Subject: [Freeswitch-users] Hooks for own call logs
In-Reply-To: <BANLkTinz0cptgEt3JQ8Eq2FDuNbgsfDdLw@mail.gmail.com>
References: <BANLkTinz0cptgEt3JQ8Eq2FDuNbgsfDdLw@mail.gmail.com>
Message-ID: <BANLkTim3mx89_p2+KmXk0-iLaBAXQC1EGA@mail.gmail.com>

It depends on what you mean by "canceled" - can you pastebin a call log of a
canceled call and give us an idea what should be happening?

Thanks,
MC

On Thu, Jun 16, 2011 at 4:59 PM, Sascha Daniels <sascha.daniels at amooma.de>wrote:

> Hi together.
>
> My dialplan is generated by a java script.
>
> I am writing call logs (not cdr) in the application that feeds the js.
>
> Everething is fine except canceled calls.
>
> I think I need a hook in the dialplan.
>
> How can I make a system call (or what ever) when a call is canceled?
>
> Any hints?
>
> Regards Sascga
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
>
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From msc at freeswitch.org  Fri Jun 17 05:46:16 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 16 Jun 2011 18:46:16 -0700
Subject: [Freeswitch-users] mod_fifo shared lag
In-Reply-To: <201106162132.47164.justlikeef@gmail.com>
References: <4DF9E16C.4090608@amooma.de>
	<201106162132.47164.justlikeef@gmail.com>
Message-ID: <BANLkTinxD_bn+P3=LTTLyqjAsyXAaes=Og@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback

There's a lag= option mentioned there and I believe that it honors multiple
queues. Can you try it out and let us know? Set a really high lag time so
that the test is completely obvious.

-MC

On Thu, Jun 16, 2011 at 6:32 PM, Rob Hutton <justlikeef at gmail.com> wrote:

> The callcenter app has a "recovery" time for each queue you are a member
> of,
> but I'm not sure about fifo.
>
> On Thursday 16 June 2011 06:56:44 Sascha Daniels wrote:
> > Hi together.
> >
> > Is there a possibility to have a shared lag over several fifos?
> >
> > In a callcenter with agents in several Queues the lag is 10sec. After a
> > call from fifo A the agent should not get a call from fifo B within
> > those 10sec.
> >
> > Basically like shared_lastcall=yes in Asterisk 1.6
> >
> > Regards
> >
> >                     Sascha
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Fri Jun 17 05:53:01 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 16 Jun 2011 18:53:01 -0700
Subject: [Freeswitch-users] how to show unregistered users
In-Reply-To: <BANLkTin0AZjiJ=FBhcrmjrW5d3BdRCYqQg@mail.gmail.com>
References: <BANLkTin0AZjiJ=FBhcrmjrW5d3BdRCYqQg@mail.gmail.com>
Message-ID: <BANLkTi=Qn3acAAq9dToXRgDp83kv1UTs_w@mail.gmail.com>

Well, if you have access to the XML config files then you have the complete
list of users. You can also parse that list out with "xml_locate directory".
Nothing that I'm aware of to tell you who isn't registered. I'm assuming
most of us will infer who is not registered by look at who is registered.

Curious - what problem are you solving by listing unregistered users?

-MC

On Thu, Jun 16, 2011 at 3:56 PM, rentmycoder rentmycoder <
rentmycoder at gmail.com> wrote:

> Hi,
>
> It may be an easy question, but I'm unable to find the answer on wiki
> or google...
>
> sofia status profie internal show only registered internal extensions...
>
> how to list all users (both reged and unreged)?
>
> like: sip show peers in Asterisk...
>
> Thanks,,,
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jaybinks at gmail.com  Fri Jun 17 06:45:37 2011
From: jaybinks at gmail.com (jay binks)
Date: Fri, 17 Jun 2011 12:45:37 +1000
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <201106162131.04859.justlikeef@gmail.com>
References: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
	<201106162131.04859.justlikeef@gmail.com>
Message-ID: <BANLkTi=5LJDDxzY2d3nirUnQGpOqoBC45Q@mail.gmail.com>

this is MY take on what Nandy is after.

A Dslam provides DSL on certain frequencies of the line..
( some of ) the other frequencies are used for voice.

my understanding is that sometimes this is split off to another device to
provide the voice,
but in this case Nandy is after a DSlam that will do the DSL part AND the
voice frequencies ( Voice signalling / audio by SIP / RTP )

I guess this is a logical question and would be quite interesting to see if
there are such devices..
sorry I dont know of any :(

Jay


On Fri, Jun 17, 2011 at 11:31 AM, Rob Hutton <justlikeef at gmail.com> wrote:

> Can you explain further what you are trying to do?  A DSLAM is a device
> that provides DSL transport, which is independent of what you run across it.
>  So normally, you have
>
> |Server | ----|network|----|DSLAM| --------[dsl line] -------[modem or
> modem/router combo]-----[network]---[Client]
>
> What the Server is (Could be Freeswitch) and what the Client is (could be a
> VOIP phone) are independent of the transport???  Onewire, Cisco, and others
> make some DSL Modem/Router combos with integrated Voip to FXS ports for the
> CPE end.
>
> If you find a DSLAM running one of the OSs that Freeswitch supports, you
> could compile Freeswitch for it, but I haven't seen one with enough CPU tp
> handle much besides the traffic.
>
> On Thursday 16 June 2011 19:03:50 Nandy Dagondon wrote:
> > hello everybody,
> >
> > i'm looking for small (24/48 ports)  IP DSLAMs that inter-connects
> directly
> > to FreeSwitch via IP.  is this already available on the market? or do we
> > still have to connect FreeSwitch via POTS splitters and FXS gateways? i
> > appreciate if you can mention some brands/models.
> >
> > tks,
> > nandy
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely

Jay
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From mays.david at gmail.com  Fri Jun 17 06:52:22 2011
From: mays.david at gmail.com (David Ma)
Date: Fri, 17 Jun 2011 10:52:22 +0800
Subject: [Freeswitch-users] Leg-A is automatically disconnected on Leg-B
 orginate failure
In-Reply-To: <BANLkTi=xYgBTPVGxVGmk9f2=bmX5X2vF=Q@mail.gmail.com>
References: <1308213151543-6482192.post@n2.nabble.com>
	<BANLkTi=xYgBTPVGxVGmk9f2=bmX5X2vF=Q@mail.gmail.com>
Message-ID: <BANLkTikhPmjN5HeeJHBnQt1ivA8zbtCnYA@mail.gmail.com>

Hi Michael,

Thanks very much for your prompt response! I appreciate the information
provided.

I was actually searching the the existence of such a variable. I was not so
luck to find it out and thereby resort to the support forum.

I've modified my code to build this parameter into my application. Will
feedback to you after verification.

Thanks again,
D.Ma

On Fri, Jun 17, 2011 at 4:51 AM, Michael Collins <msc at freeswitch.org> wrote:

> How about setting this?
> http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
>
> -MC
>
>
> On Thu, Jun 16, 2011 at 1:32 AM, dma <mays.david at gmail.com> wrote:
>
>> I am creating a call-back solution. After leg-A answers, I originate leg-B
>> call. After receiving SIP 183 from Leg-B, I bridge the 2 legs. However, in
>> some cases, leg-A is automatically disconnected by FreeSwitch on leg-B
>> failure, for example, DESTINATION_OUT_OF_ORDER. The application is not
>> given
>> a chance to handle leg-B failure event. This should not be a correct
>> scenario because I never set "hangup-after-bridge", which is false by
>> default.
>>
>> The right way should be, FreeSwitch doesn't hang up leg-A automatically,
>> but
>> give a chance for the application to decide what to do.
>>
>> Please see the logs below:
>>
>> =================================================
>>
>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
>> string 0 = [origination_caller_id_number=03996563750914]
>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
>> string 1 = [originate_timeout=30]
>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
>> string 2 = [ccd_session_id=20110610105829676824]
>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
>> string 3 = [sip_cid_type=pid]
>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
>> string 4 = [privacy=yes]
>> 2011-06-10 11:09:55.370506 [NOTICE] switch_channel.c:808 New Channel
>> sofia/external/03996590031055 at 203.208.207.212
>> [ea57b74b-a8c2-4fea-9683-98054dc03a79]
>> 2011-06-10 11:09:55.370506 [DEBUG] mod_sofia.c:4129
>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_NEW ->
>> CS_INIT
>> 2011-06-10 11:09:55.370506 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>> 2011-06-10 11:09:55.370506 [DEBUG] switch_core_state_machine.c:320
>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>> CS_INIT
>> 2011-06-10 11:09:55.370506 [DEBUG] switch_core_state_machine.c:356
>> (sofia/external/03996590031055 at 203.208.207.212) State INIT
>> 2011-06-10 11:09:55.370506 [DEBUG] mod_sofia.c:84
>> sofia/external/03996590031055 at 203.208.207.212 SOFIA INIT
>> send 984 bytes to udp/[203.208.207.212]:5060 at 03:09:55.477997:
>>   ------------------------------------------------------------------------
>>   INVITE sip:03996590031055 at 203.208.207.212 SIP/2.0
>>   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKXjQ7eFpKypy5D
>>   Max-Forwards: 70
>>   From: "" &lt;sip:03996563750914 at 202.73.56.46&gt;;tag=H4ZN43UH9S9Qc
>>   To: &lt;sip:03996590031055 at 203.208.207.212&gt;
>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>   CSeq: 13501633 INVITE
>>   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
>>   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY
>>   Supported: timer, precondition, path, replaces
>>   Allow-Events: talk, hold, refer
>>   Privacy: none
>>   Content-Type: application/sdp
>>   Content-Disposition: session
>>   Content-Length: 204
>>   X-FS-Support: update_display
>>   P-Asserted-Identity: "" &lt;sip:03996563750914 at 202.73.56.46&gt;
>>
>>   v=0
>>   o=FreeSWITCH 1307645395 1307645396 IN IP4 202.73.56.46
>>   s=FreeSWITCH
>>   c=IN IP4 202.73.56.46
>>   t=0 0
>>   m=audio 30000 RTP/AVP 18 3 101 13
>>   a=rtpmap:101 telephone-event/8000
>>   a=fmtp:101 0-16
>>   a=ptime:20
>>   ------------------------------------------------------------------------
>> 2011-06-10 11:09:55.371674 [DEBUG] mod_sofia.c:124
>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_INIT ->
>> CS_ROUTING
>> 2011-06-10 11:09:55.371674 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>> 2011-06-10 11:09:55.371674 [DEBUG] sofia.c:4646 Channel
>> sofia/external/03996590031055 at 203.208.207.212 entering state [calling][0]
>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:356
>> (sofia/external/03996590031055 at 203.208.207.212) State INIT going to sleep
>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:320
>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>> CS_ROUTING
>> 2011-06-10 11:09:55.373478 [DEBUG] switch_channel.c:1657
>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change DOWN ->
>> RINGING
>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:359
>> (sofia/external/03996590031055 at 203.208.207.212) State ROUTING
>> 2011-06-10 11:09:55.373478 [DEBUG] mod_sofia.c:147
>> sofia/external/03996590031055 at 203.208.207.212 SOFIA ROUTING
>> 2011-06-10 11:09:55.373478 [DEBUG] switch_ivr_originate.c:66
>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_ROUTING
>> ->
>> CS_CONSUME_MEDIA
>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:359
>> (sofia/external/03996590031055 at 203.208.207.212) State ROUTING going to
>> sleep
>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:320
>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>> CS_CONSUME_MEDIA
>> 2011-06-10 11:09:55.374485 [DEBUG] switch_core_state_machine.c:378
>> (sofia/external/03996590031055 at 203.208.207.212) State CONSUME_MEDIA
>> 2011-06-10 11:09:55.374485 [DEBUG] switch_core_state_machine.c:378
>> (sofia/external/03996590031055 at 203.208.207.212) State CONSUME_MEDIA going
>> to
>> sleep
>> recv 307 bytes from udp/[203.208.207.212]:5060 at 03:09:55.485130:
>>   ------------------------------------------------------------------------
>>   SIP/2.0 100 Trying
>>   Via: SIP/2.0/UDP
>> 10.1.1.46:5080;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>   To: &lt;sip:03996590031055 at 203.208.207.212&gt;
>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>   CSeq: 13501633 INVITE
>>   Content-Length:     0
>>
>>   ------------------------------------------------------------------------
>> recv 669 bytes from udp/[203.208.207.212]:5060 at 03:09:56.677788:
>>   ------------------------------------------------------------------------
>>   SIP/2.0 183 Session Progress
>>   Via: SIP/2.0/UDP
>> 10.1.1.46:5080;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>   To:
>> &lt;sip:03996590031055 at 203.208.207.212
>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>   CSeq: 13501633 INVITE
>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
>> SUBSCRIBE, UPDATE
>>   Content-Type: application/sdp
>>   Content-Length:   189
>>
>>   v=0
>>   o=- 421265648 1 IN IP4 203.208.207.219
>>   s=session
>>   c=IN IP4 203.208.207.196
>>   t=0 0
>>   m=audio 30792 RTP/AVP 18 101
>>   a=rtpmap:18 G729/8000
>>   a=rtpmap:101 telephone-event/8000
>>   a=sendrecv
>>   ------------------------------------------------------------------------
>> 2011-06-10 11:09:56.571839 [INFO] sofia.c:729
>> sofia/external/03996590031055 at 203.208.207.212 Update Callee ID to
>> "Outbound
>> Call" <03996590031055>
>> 2011-06-10 11:09:56.571839 [DEBUG] sofia.c:4646 Channel
>> sofia/external/03996590031055 at 203.208.207.212 entering state
>> [proceeding][183]
>> 2011-06-10 11:09:56.571839 [DEBUG] sofia.c:4657 Remote SDP:
>> v=0
>> o=- 421265648 1 IN IP4 203.208.207.219
>> s=session
>> c=IN IP4 203.208.207.196
>> t=0 0
>> m=audio 30792 RTP/AVP 18 101
>> a=rtpmap:18 G729/8000
>> a=rtpmap:101 telephone-event/8000
>>
>> 2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:4467 Audio Codec Compare
>> [G729:18:8000:20:8000]/[G729:18:8000:20:8000]
>> 2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:2757 Set Codec
>> sofia/external/03996590031055 at 203.208.207.212 G729/8000 20 ms 160 samples
>> 8000 bits
>> 2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:4565 Set 2833 dtmf send
>> payload to 101
>> 2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:2987 AUDIO RTP
>> [sofia/external/03996590031055 at 203.208.207.212] 10.1.1.46 port 30000 ->
>> 203.208.207.196 port 30792 codec: 18 ms: 20
>> 2011-06-10 11:09:56.571839 [DEBUG] switch_rtp.c:1607 Starting timer [soft]
>> 160 bytes per 20ms
>> 2011-06-10 11:09:56.573731 [DEBUG] sofia_glue.c:3228 Set 2833 dtmf send
>> payload to 101
>> 2011-06-10 11:09:56.573731 [DEBUG] sofia_glue.c:3233 Set 2833 dtmf receive
>> payload to 101
>> 2011-06-10 11:09:56.573731 [NOTICE] sofia_glue.c:3680 Pre-Answer
>> sofia/external/03996590031055 at 203.208.207.212!
>> 2011-06-10 11:09:56.573731 [DEBUG] switch_channel.c:2627
>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change RINGING
>> ->
>> EARLY
>> 2011-06-10 11:09:56.573731 [DEBUG] switch_ivr_originate.c:3408 Originate
>> Resulted in Success: [sofia/external/03996590031055 at 203.208.207.212]
>> 2011-06-10 11:09:56.573731 [DEBUG] mod_commands.c:3205
>> (sofia/external/03996590031055 at 203.208.207.212) State Change
>> CS_CONSUME_MEDIA -> CS_EXECUTE
>> 2011-06-10 11:09:56.573731 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>> 2011-06-10 11:09:56.575262 [DEBUG] switch_core_state_machine.c:320
>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>> CS_EXECUTE
>> 2011-06-10 11:09:56.575262 [DEBUG] switch_core_state_machine.c:366
>> (sofia/external/03996590031055 at 203.208.207.212) State EXECUTE
>> 2011-06-10 11:09:56.575262 [DEBUG] mod_sofia.c:240
>> sofia/external/03996590031055 at 203.208.207.212 SOFIA EXECUTE
>> 2011-06-10 11:09:56.575262 [DEBUG] switch_core_state_machine.c:157
>> sofia/external/03996590031055 at 203.208.207.212 Standard EXECUTE
>> EXECUTE sofia/external/03996590031055 at 203.208.207.212 park()
>> 2011-06-10 11:09:56.618770 [DEBUG] switch_rtp.c:2933 Correct ip/port
>> confirmed.
>> recv 669 bytes from udp/[203.208.207.212]:5060 at 03:10:01.562021:
>>   ------------------------------------------------------------------------
>>   SIP/2.0 183 Session Progress
>>   Via: SIP/2.0/UDP
>> 10.1.1.46:5080;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>   To:
>> &lt;sip:03996590031055 at 203.208.207.212
>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>   CSeq: 13501633 INVITE
>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
>> SUBSCRIBE, UPDATE
>>   Content-Type: application/sdp
>>   Content-Length:   189
>>
>>   v=0
>>   o=- 421265648 2 IN IP4 203.208.207.219
>>   s=session
>>   c=IN IP4 203.208.207.196
>>   t=0 0
>>   m=audio 30792 RTP/AVP 18 101
>>   a=rtpmap:18 G729/8000
>>   a=rtpmap:101 telephone-event/8000
>>   a=sendrecv
>>   ------------------------------------------------------------------------
>> 2011-06-10 11:10:01.455944 [INFO] sofia.c:729
>> sofia/external/03996590031055 at 203.208.207.212 Update Callee ID to
>> "03996590031055" <03996590031055>
>> 2011-06-10 11:10:01.455944 [DEBUG] sofia.c:4641 Channel
>> sofia/external/03996590031055 at 203.208.207.212 skipping state
>> [proceeding][183]
>> recv 669 bytes from udp/[203.208.207.212]:5060 at 03:10:01.563178:
>>   ------------------------------------------------------------------------
>>   SIP/2.0 183 Session Progress
>>   Via: SIP/2.0/UDP
>> 10.1.1.46:5080;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>   To:
>> &lt;sip:03996590031055 at 203.208.207.212
>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>   CSeq: 13501633 INVITE
>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
>> SUBSCRIBE, UPDATE
>>   Content-Type: application/sdp
>>   Content-Length:   189
>>
>>   v=0
>>   o=- 421265648 3 IN IP4 203.208.207.219
>>   s=session
>>   c=IN IP4 203.208.207.196
>>   t=0 0
>>   m=audio 30792 RTP/AVP 18 101
>>   a=rtpmap:18 G729/8000
>>   a=rtpmap:101 telephone-event/8000
>>   a=sendrecv
>>   ------------------------------------------------------------------------
>> 2011-06-10 11:10:01.457169 [DEBUG] sofia.c:4641 Channel
>> sofia/external/03996590031055 at 203.208.207.212 skipping state
>> [proceeding][183]
>> recv 749 bytes from udp/[203.208.207.212]:5060 at 03:10:27.365284:
>>   ------------------------------------------------------------------------
>>   SIP/2.0 200 Ok
>>   Via: SIP/2.0/UDP
>> 10.1.1.46:5080;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>   To:
>> &lt;sip:03996590031055 at 203.208.207.212
>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>   CSeq: 13501633 INVITE
>>   Contact: <sip:203.208.207.212:5060>
>>   Allow-Events: refer
>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
>> SUBSCRIBE, UPDATE
>>   Content-Type: application/sdp
>>   Supported: 100rel, timer, replaces
>>   Content-Length:   189
>>
>>   v=0
>>   o=- 421265648 4 IN IP4 203.208.207.219
>>   s=session
>>   c=IN IP4 203.208.207.196
>>   t=0 0
>>   m=audio 30792 RTP/AVP 18 101
>>   a=rtpmap:18 G729/8000
>>   a=rtpmap:101 telephone-event/8000
>>   a=sendrecv
>>   ------------------------------------------------------------------------
>> 2011-06-10 11:10:27.258092 [DEBUG] sofia.c:4646 Channel
>> sofia/external/03996590031055 at 203.208.207.212 entering state
>> [completing][200]
>> 2011-06-10 11:10:27.258092 [DEBUG] sofia.c:4657 Remote SDP:
>> v=0
>> o=- 421265648 4 IN IP4 203.208.207.219
>> s=session
>> c=IN IP4 203.208.207.196
>> t=0 0
>> m=audio 30792 RTP/AVP 18 101
>> a=rtpmap:18 G729/8000
>> a=rtpmap:101 telephone-event/8000
>>
>> send 405 bytes to udp/[203.208.207.212]:5060 at 03:10:27.366562:
>>   ------------------------------------------------------------------------
>>   ACK sip:203.208.207.212:5060 SIP/2.0
>>   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKyUg0ga7pUZmrS
>>   Max-Forwards: 70
>>   From: "" &lt;sip:03996563750914 at 202.73.56.46&gt;;tag=H4ZN43UH9S9Qc
>>   To:
>> &lt;sip:03996590031055 at 203.208.207.212
>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>   CSeq: 13501633 ACK
>>   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
>>   Content-Length: 0
>>
>>   ------------------------------------------------------------------------
>> 2011-06-10 11:10:27.260267 [DEBUG] sofia.c:4646 Channel
>> sofia/external/03996590031055 at 203.208.207.212 entering state [ready][200]
>> 2011-06-10 11:10:27.260267 [DEBUG] switch_channel.c:2782
>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change EARLY ->
>> ACTIVE
>> 2011-06-10 11:10:27.260267 [NOTICE] sofia.c:5175 Channel
>> [sofia/external/03996590031055 at 203.208.207.212] has been answered
>> 2011-06-10 11:10:27.264178 [DEBUG] switch_scheduler.c:214 Added task 23
>> switch_ivr_schedule_hangup (ea57b74b-a8c2-4fea-9683-98054dc03a79) to run
>> at
>> 1307676927
>> 2011-06-10 11:10:27.265202 [DEBUG] switch_core_session.c:954 Send signal
>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>> 2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971 variable
>> string 0 = [origination_caller_id_number=03996590031055]
>> 2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971 variable
>> string 1 = [ccd_session_id=20110610105829676824]
>> 2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971 variable
>> string 2 = [sip_cid_type=pid]
>> 2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971 variable
>> string 3 = [privacy=yes]
>> 2011-06-10 11:10:27.266218 [NOTICE] switch_channel.c:808 New Channel
>> sofia/external/03996563750914 at 203.208.207.212
>> [30228d2b-756a-4a98-871d-db63a2955b52]
>> 2011-06-10 11:10:27.266218 [DEBUG] mod_sofia.c:4129
>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_NEW ->
>> CS_INIT
>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:320
>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>> CS_INIT
>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:356
>> (sofia/external/03996563750914 at 203.208.207.212) State INIT
>> 2011-06-10 11:10:27.266218 [DEBUG] mod_sofia.c:84
>> sofia/external/03996563750914 at 203.208.207.212 SOFIA INIT
>> 2011-06-10 11:10:27.266218 [DEBUG] mod_sofia.c:124
>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_INIT ->
>> CS_ROUTING
>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:356
>> (sofia/external/03996563750914 at 203.208.207.212) State INIT going to sleep
>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:320
>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>> CS_ROUTING
>> 2011-06-10 11:10:27.266218 [DEBUG] switch_channel.c:1657
>> (sofia/external/03996563750914 at 203.208.207.212) Callstate Change DOWN ->
>> RINGING
>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:359
>> (sofia/external/03996563750914 at 203.208.207.212) State ROUTING
>> send 984 bytes to udp/[203.208.207.212]:5060 at 03:10:27.373220:
>>   ------------------------------------------------------------------------
>>   INVITE sip:03996563750914 at 203.208.207.212 SIP/2.0
>>   Via: SIP/2.0/UDP
>> 202.73.56.46:5080;rport;branch=z9hG4bKZ49rj5Qtr8aBN2011-06-10
>> 11:10:27.266218 [DEBUG] mod_sofia.c:147
>> sofia/external/03996563750914 at 203.208.207.212 SOFIA ROUTING
>>
>>   Max-Forwards: 70
>>   From: "" &lt;sip:03996590031055 at 202.73.56.46&gt;;tag=jDSe6ycN62Zar
>>   To: &lt;sip:03996563750914 at 203.208.207.212&gt;
>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>   CSeq: 13501649 INVITE
>>   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
>>   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY
>>   Supported: timer, precondition, path, replaces
>>   Allow-Events: talk, hold, refer
>>   Privacy: none
>>   Content-Type: application/sdp
>>   Content-Disposition: session
>>   Content-Length: 204
>>   X-FS-Support: update_display
>>   P-Asserted-Identity: "" &lt;sip:03996590031055 at 202.73.56.46&gt;
>>
>>   v=0
>>   o=FreeSWITCH 1307646863 1307646864 IN IP4 202.73.56.46
>>   s=FreeSWITCH
>>   c=IN IP4 202.73.56.46
>>   t=0 0
>>   m=audio 28564 RTP/AVP 18 3 101 13
>>   a=rtpmap:101 telephone-event/8000
>>   a=fmtp:101 0-16
>>   a=ptime:20
>>   ------------------------------------------------------------------------
>> 2011-06-10 11:10:27.267630 [DEBUG] sofia.c:4646 Channel
>> sofia/external/03996563750914 at 203.208.207.212 entering state [calling][0]
>> 2011-06-10 11:10:27.267630 [DEBUG] switch_ivr_originate.c:66
>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_ROUTING
>> ->
>> CS_CONSUME_MEDIA
>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:359
>> (sofia/external/03996563750914 at 203.208.207.212) State ROUTING going to
>> sleep
>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:320
>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>> CS_CONSUME_MEDIA
>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:378
>> (sofia/external/03996563750914 at 203.208.207.212) State CONSUME_MEDIA
>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:378
>> (sofia/external/03996563750914 at 203.208.207.212) State CONSUME_MEDIA going
>> to
>> sleep
>> recv 307 bytes from udp/[203.208.207.212]:5060 at 03:10:27.378710:
>>   ------------------------------------------------------------------------
>>   SIP/2.0 100 Trying
>>   Via: SIP/2.0/UDP
>> 10.1.1.46:5080;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>   To: &lt;sip:03996563750914 at 203.208.207.212&gt;
>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>   CSeq: 13501649 INVITE
>>   Content-Length:     0
>>
>>   ------------------------------------------------------------------------
>> 2011-06-10 11:10:27.278484 [DEBUG] switch_ivr.c:563
>> sofia/external/03996590031055 at 203.208.207.212 Command Execute
>> playback(/usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav)
>> EXECUTE sofia/external/03996590031055 at 203.208.207.212
>> playback(/usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav)
>> 2011-06-10 11:10:27.278484 [DEBUG] switch_core_file.c:176 File
>> /usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav sample rate
>> 11025
>> doesn't match requested rate 8000
>> 2011-06-10 11:10:27.278484 [WARNING] switch_core_file.c:189 File has 2
>> channels, muxing to mono will occur.
>> 2011-06-10 11:10:27.278484 [DEBUG] switch_ivr_play_say.c:1244 Codec
>> Activated L16 at 8000hz 2 channels 20ms
>> 2011-06-10 11:10:27.298764 [INFO] mod_com_g729.c:119 ENCODER CREATE -
>> 0x2aaab00310c0 0x2aaab00b20c0
>> recv 666 bytes from udp/[203.208.207.212]:5060 at 03:10:28.618472:
>>   ------------------------------------------------------------------------
>>   SIP/2.0 183 Session Progress
>>   Via: SIP/2.0/UDP
>> 10.1.1.46:5080;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>   To:
>> &lt;sip:03996563750914 at 203.208.207.212
>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>   CSeq: 13501649 INVITE
>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
>> SUBSCRIBE, UPDATE
>>   Content-Type: application/sdp
>>   Content-Length:   186
>>
>>   v=0
>>   o=- 131082 1 IN IP4 203.208.207.218
>>   s=session
>>   c=IN IP4 203.208.207.195
>>   t=0 0
>>   m=audio 45002 RTP/AVP 18 101
>>   a=rtpmap:18 G729/8000
>>   a=rtpmap:101 telephone-event/8000
>>   a=sendrecv
>>   ------------------------------------------------------------------------
>> 2011-06-10 11:10:28.513195 [INFO] sofia.c:729
>> sofia/external/03996563750914 at 203.208.207.212 Update Callee ID to
>> "Outbound
>> Call" <03996563750914>
>> 2011-06-10 11:10:28.513195 [DEBUG] sofia.c:4646 Channel
>> sofia/external/03996563750914 at 203.208.207.212 entering state
>> [proceeding][183]
>> 2011-06-10 11:10:28.513195 [DEBUG] sofia.c:4657 Remote SDP:
>> v=0
>> o=- 131082 1 IN IP4 203.208.207.218
>> s=session
>> c=IN IP4 203.208.207.195
>> t=0 0
>> m=audio 45002 RTP/AVP 18 101
>> a=rtpmap:18 G729/8000
>> a=rtpmap:101 telephone-event/8000
>>
>> 2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:4467 Audio Codec Compare
>> [G729:18:8000:20:8000]/[G729:18:8000:20:8000]
>> 2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:2757 Set Codec
>> sofia/external/03996563750914 at 203.208.207.212 G729/8000 20 ms 160 samples
>> 8000 bits
>> 2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:4565 Set 2833 dtmf send
>> payload to 101
>> 2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:2987 AUDIO RTP
>> [sofia/external/03996563750914 at 203.208.207.212] 10.1.1.46 port 28564 ->
>> 203.208.207.195 port 45002 codec: 18 ms: 20
>> 2011-06-10 11:10:28.513195 [DEBUG] switch_rtp.c:1607 Starting timer [soft]
>> 160 bytes per 20ms
>> 2011-06-10 11:10:28.514938 [DEBUG] sofia_glue.c:3228 Set 2833 dtmf send
>> payload to 101
>> 2011-06-10 11:10:28.514938 [DEBUG] sofia_glue.c:3233 Set 2833 dtmf receive
>> payload to 101
>> 2011-06-10 11:10:28.514938 [NOTICE] sofia_glue.c:3680 Pre-Answer
>> sofia/external/03996563750914 at 203.208.207.212!
>> 2011-06-10 11:10:28.514938 [DEBUG] switch_channel.c:2627
>> (sofia/external/03996563750914 at 203.208.207.212) Callstate Change RINGING
>> ->
>> EARLY
>> 2011-06-10 11:10:28.514938 [DEBUG] switch_ivr_originate.c:3408 Originate
>> Resulted in Success: [sofia/external/03996563750914 at 203.208.207.212]
>> 2011-06-10 11:10:28.514938 [DEBUG] mod_commands.c:3205
>> (sofia/external/03996563750914 at 203.208.207.212) State Change
>> CS_CONSUME_MEDIA -> CS_EXECUTE
>> 2011-06-10 11:10:28.514938 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>> 2011-06-10 11:10:28.515884 [DEBUG] switch_core_state_machine.c:320
>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>> CS_EXECUTE
>> 2011-06-10 11:10:28.515884 [DEBUG] switch_core_state_machine.c:366
>> (sofia/external/03996563750914 at 203.208.207.212) State EXECUTE
>> 2011-06-10 11:10:28.515884 [DEBUG] mod_sofia.c:240
>> sofia/external/03996563750914 at 203.208.207.212 SOFIA EXECUTE
>> 2011-06-10 11:10:28.515884 [DEBUG] switch_core_state_machine.c:157
>> sofia/external/03996563750914 at 203.208.207.212 Standard EXECUTE
>> EXECUTE sofia/external/03996563750914 at 203.208.207.212 park()
>> 2011-06-10 11:10:28.516887 [DEBUG] switch_core_session.c:954 Send signal
>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>> 2011-06-10 11:10:28.517896 [DEBUG] switch_ivr_bridge.c:1480
>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_EXECUTE
>> ->
>> CS_HIBERNATE
>> 2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>> 2011-06-10 11:10:28.517896 [DEBUG] switch_ivr_bridge.c:1482
>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_EXECUTE
>> ->
>> CS_HIBERNATE
>> 2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>> 2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:771 Send signal
>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>> 2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:771 Send signal
>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>> 2011-06-10 11:10:28.519223 [DEBUG] switch_ivr_play_say.c:1581 done playing
>> file
>> 2011-06-10 11:10:28.519223 [DEBUG] switch_ivr.c:563
>> sofia/external/03996590031055 at 203.208.207.212 Command Execute
>> playback(tone_stream://%(2000,4000,440,480);loops=10)
>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:366
>> (sofia/external/03996563750914 at 203.208.207.212) State EXECUTE going to
>> sleep
>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:320
>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>> CS_HIBERNATE
>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:381
>> (sofia/external/03996563750914 at 203.208.207.212) State HIBERNATE
>> 2011-06-10 11:10:28.519223 [DEBUG] mod_sofia.c:221
>> sofia/external/03996563750914 at 203.208.207.212 SOFIA HIBERNATE
>> 2011-06-10 11:10:28.519223 [DEBUG] switch_ivr_bridge.c:731
>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_HIBERNATE
>> ->
>> CS_RESET
>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:381
>> (sofia/external/03996563750914 at 203.208.207.212) State HIBERNATE going to
>> sleep
>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:320
>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>> CS_RESET
>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:362
>> (sofia/external/03996563750914 at 203.208.207.212) State RESET
>> 2011-06-10 11:10:28.519223 [DEBUG] mod_sofia.c:165
>> sofia/external/03996563750914 at 203.208.207.212 SOFIA RESET
>> 2011-06-10 11:10:28.519223 [DEBUG] switch_ivr_bridge.c:716
>> sofia/external/03996563750914 at 203.208.207.212 CUSTOM RESET
>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:66
>> sofia/external/03996563750914 at 203.208.207.212 Standard RESET
>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:362
>> (sofia/external/03996563750914 at 203.208.207.212) State RESET going to
>> sleep
>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_session.c:709 Send signal
>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>> recv 666 bytes from udp/[203.208.207.212]:5060 at 03:10:30.540814:
>>   ------------------------------------------------------------------------
>>   SIP/2.0 183 Session Progress
>>   Via: SIP/2.0/UDP
>> 10.1.1.46:5080;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>   To:
>> &lt;sip:03996563750914 at 203.208.207.212
>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>   CSeq: 13501649 INVITE
>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
>> SUBSCRIBE, UPDATE
>>   Content-Type: application/sdp
>>   Content-Length:   186
>>
>>   v=0
>>   o=- 131082 2 IN IP4 203.208.207.218
>>   s=session
>>   c=IN IP4 203.208.207.195
>>   t=0 0
>>   m=audio 45002 RTP/AVP 18 101
>>   a=rtpmap:18 G729/8000
>>   a=rtpmap:101 telephone-event/8000
>>   a=sendrecv
>>   ------------------------------------------------------------------------
>> 2011-06-10 11:10:30.435309 [INFO] sofia.c:729
>> sofia/external/03996563750914 at 203.208.207.212 Update Callee ID to
>> "03996563750914" <03996563750914>
>> 2011-06-10 11:10:30.435309 [DEBUG] sofia.c:4641 Channel
>> sofia/external/03996563750914 at 203.208.207.212 skipping state
>> [proceeding][183]
>> 2011-06-10 11:10:30.691404 [WARNING] switch_core_session.c:1940 Cannot
>> execute app 'playback' media required on an outbound channel that does not
>> have media established
>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:366
>> (sofia/external/03996590031055 at 203.208.207.212) State EXECUTE going to
>> sleep
>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>> CS_HIBERNATE
>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:381
>> (sofia/external/03996590031055 at 203.208.207.212) State HIBERNATE
>> 2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:221
>> sofia/external/03996590031055 at 203.208.207.212 SOFIA HIBERNATE
>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:731
>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_HIBERNATE
>> ->
>> CS_RESET
>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:381
>> (sofia/external/03996590031055 at 203.208.207.212) State HIBERNATE going to
>> sleep
>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>> CS_RESET
>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:362
>> (sofia/external/03996590031055 at 203.208.207.212) State RESET
>> 2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:165
>> sofia/external/03996590031055 at 203.208.207.212 SOFIA RESET
>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:716
>> sofia/external/03996590031055 at 203.208.207.212 CUSTOM RESET
>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:723
>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_RESET ->
>> CS_SOFT_EXECUTE
>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:362
>> (sofia/external/03996590031055 at 203.208.207.212) State RESET going to
>> sleep
>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>> CS_SOFT_EXECUTE
>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:372
>> (sofia/external/03996590031055 at 203.208.207.212) State SOFT_EXECUTE
>> 2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:558 SOFIA SOFT_EXECUTE
>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:741
>> sofia/external/03996590031055 at 203.208.207.212 CUSTOM SOFT_EXECUTE
>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:761
>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_RESET ->
>> CS_SOFT_EXECUTE
>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>> CS_SOFT_EXECUTE
>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:372
>> (sofia/external/03996563750914 at 203.208.207.212) State SOFT_EXECUTE
>> 2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:558 SOFIA SOFT_EXECUTE
>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:741
>> sofia/external/03996563750914 at 203.208.207.212 CUSTOM SOFT_EXECUTE
>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:204
>> sofia/external/03996563750914 at 203.208.207.212 Standard SOFT_EXECUTE
>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:372
>> (sofia/external/03996563750914 at 203.208.207.212) State SOFT_EXECUTE going
>> to
>> sleep
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2535
>> (sofia/external/03996563750914 at 203.208.207.212) Callstate Change EARLY ->
>> HANGUP
>> 2011-06-10 11:10:30.711798 [NOTICE] switch_ivr_originate.c:1045 Hangup
>> sofia/external/03996563750914 at 203.208.207.212 [CS_SOFT_EXECUTE]
>> [DESTINATION_OUT_OF_ORDER]
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2551 Send signal
>> sofia/external/03996563750914 at 203.208.207.212 [KILL]
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>> CS_HANGUP
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2535
>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change ACTIVE
>> ->
>> HANGUP
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
>> (sofia/external/03996563750914 at 203.208.207.212) State HANGUP
>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:457 Channel
>> sofia/external/03996563750914 at 203.208.207.212 hanging up, cause:
>> DESTINATION_OUT_OF_ORDER
>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:510 Sending CANCEL to
>> sofia/external/03996563750914 at 203.208.207.212
>> send 390 bytes to udp/[203.208.207.212]:5060 at 03:10:30.818391:
>>   ------------------------------------------------------------------------
>>   CANCEL sip:03996563750914 at 203.208.207.212 SIP/2.0
>>   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKZ49rj5Qtr8aBN
>>   Max-Forwards: 70
>>   From: "" &lt;sip:03996590031055 at 202.73.56.46&gt;;tag=jDSe6ycN62Zar
>>   To: &lt;sip:03996563750914 at 203.208.207.212&gt;
>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>   CSeq: 13501649 CANCEL
>>   Reason: Q.850;cause=27;text="DESTINATION_OUT_OF_ORDER"
>>   Content-Length: 0
>>
>>   ------------------------------------------------------------------------
>> 2011-06-10 11:10:30.711798 [NOTICE] switch_ivr_bridge.c:772 Hangup
>> sofia/external/03996590031055 at 203.208.207.212 [CS_SOFT_EXECUTE]
>> [ORIGINATOR_CANCEL]
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:46
>> sofia/external/03996563750914 at 203.208.207.212 Standard HANGUP, cause:
>> DESTINATION_OUT_OF_ORDER
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
>> (sofia/external/03996563750914 at 203.208.207.212) State HANGUP going to
>> sleep
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:351
>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_HANGUP ->
>> CS_REPORTING
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>> CS_REPORTING
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:617
>> (sofia/external/03996563750914 at 203.208.207.212) State REPORTING
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:53
>> sofia/external/03996563750914 at 203.208.207.212 Standard REPORTING, cause:
>> DESTINATION_OUT_OF_ORDER
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:617
>> (sofia/external/03996563750914 at 203.208.207.212) State REPORTING going to
>> sleep
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2551 Send signal
>> sofia/external/03996590031055 at 203.208.207.212 [KILL]
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:372
>> (sofia/external/03996590031055 at 203.208.207.212) State SOFT_EXECUTE going
>> to
>> sleep
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>> CS_HANGUP
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:345
>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_REPORTING
>> ->
>> CS_DESTROY
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1288 Session 40
>> (sofia/external/03996563750914 at 203.208.207.212) Locked, Waiting on
>> external
>> entities
>> 2011-06-10 11:10:30.711798 [NOTICE] switch_core_session.c:1306 Session 40
>> (sofia/external/03996563750914 at 203.208.207.212) Ended
>> 2011-06-10 11:10:30.711798 [NOTICE] switch_core_session.c:1308 Close
>> Channel
>> sofia/external/03996563750914 at 203.208.207.212 [CS_DESTROY]
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:449
>> (sofia/external/03996563750914 at 203.208.207.212) Callstate Change HANGUP
>> ->
>> DOWN
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:452
>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>> CS_DESTROY
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:462
>> (sofia/external/03996563750914 at 203.208.207.212) State DESTROY
>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:362
>> sofia/external/03996563750914 at 203.208.207.212 SOFIA DESTROY
>> 2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
>> 0x2aaaac013028 (nil)
>> 2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
>> 0x2aaaac013028 (nil)
>> 2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
>> 0x2aaaac013088 (nil)
>> 2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
>> 0x2aaaac013088 (nil)
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:60
>> sofia/external/03996563750914 at 203.208.207.212 Standard DESTROY
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:462
>> (sofia/external/03996563750914 at 203.208.207.212) State DESTROY going to
>> sleep
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
>> (sofia/external/03996590031055 at 203.208.207.212) State HANGUP
>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:457 Channel
>> sofia/external/03996590031055 at 203.208.207.212 hanging up, cause:
>> ORIGINATOR_CANCEL
>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:500 Sending BYE to
>> sofia/external/03996590031055 at 203.208.207.212
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:46
>> sofia/external/03996590031055 at 203.208.207.212 Standard HANGUP, cause:
>> ORIGINATOR_CANCEL
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
>> (sofia/external/03996590031055 at 203.208.207.212) State HANGUP going to
>> sleep
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:351
>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_HANGUP ->
>> CS_REPORTING
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>> CS_REPORTING
>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:617
>> (sofia/external/03996590031055 at 203.208.207.212) State REPORTING
>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:53
>> sofia/external/03996590031055 at 203.208.207.212 Standard REPORTING, cause:
>> ORIGINATOR_CANCEL
>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:617
>> (sofia/external/03996590031055 at 203.208.207.212) State REPORTING going to
>> sleep
>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:345
>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_REPORTING
>> ->
>> CS_DESTROY
>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_session.c:1288 Session 39
>> (sofia/external/03996590031055 at 203.208.207.212) Locked, Waiting on
>> external
>> entities
>> 2011-06-10 11:10:30.714001 [NOTICE] switch_core_session.c:1306 Session 39
>> (sofia/external/03996590031055 at 203.208.207.212) Ended
>> 2011-06-10 11:10:30.714001 [NOTICE] switch_core_session.c:1308 Close
>> Channel
>> sofia/external/03996590031055 at 203.208.207.212 [CS_DESTROY]
>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:449
>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change HANGUP
>> ->
>> DOWN
>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:452
>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>> CS_DESTROY
>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:462
>> (sofia/external/03996590031055 at 203.208.207.212) State DESTROY
>> 2011-06-10 11:10:30.714001 [DEBUG] mod_sofia.c:362
>> sofia/external/03996590031055 at 203.208.207.212 SOFIA DESTROY
>> 2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
>> 0x2aaab0031060 (nil)
>> 2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
>> 0x2aaab0031060 (nil)
>> 2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
>> 0x2aaab00310c0 0x2aaab00b20c0
>> 2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
>> 0x2aaab00310c0 (nil)
>> send 662 bytes to udp/[203.208.207.212]:5060 at 03:10:30.820530:
>>   ------------------------------------------------------------------------
>>   BYE sip:203.208.207.212:5060 SIP/2.0
>>   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bK0D3Hm08XNH1Xg
>>   Max-Forwards: 70
>>   From: "" &lt;sip:03996563750914 at 202.73.56.46&gt;;tag=H4ZN43UH9S9Qc
>>   To:
>> &lt;sip:03996590031055 at 203.208.207.212
>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>   CSeq: 13501634 BYE
>>   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
>>   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY
>>   Supported: timer, precondition, path, replaces
>>   Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL"
>>   Content-Length: 0
>>
>>   ------------------------------------------------------------------------
>> 2011-06-10 11:10:30.715878 [INFO] mod_com_g729.c:83 ENCODER DESTROY -
>> 0x2aaab00310c0 0x2aaab00b20c0
>> 2011-06-10 11:10:30.715878 [DEBUG] switch_core_state_machine.c:60
>> sofia/external/03996590031055 at 203.208.207.212 Standard DESTROY
>> 2011-06-10 11:10:30.715878 [DEBUG] switch_core_state_machine.c:462
>> (sofia/external/03996590031055 at 203.208.207.212) State DESTROY going to
>> sleep
>> recv 383 bytes from udp/[203.208.207.212]:5060 at 03:10:30.823302:
>>   ------------------------------------------------------------------------
>>   SIP/2.0 200 Ok
>>   Via: SIP/2.0/UDP
>> 10.1.1.46:5080;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>   To:
>> &lt;sip:03996563750914 at 203.208.207.212
>> &gt;;tag=2QGB951HCR30000E1D00000u00000001QXU3LU
>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>   CSeq: 13501649 CANCEL
>>   Contact: <sip:203.208.207.212:5060>
>>   Content-Length:     0
>>
>>   ------------------------------------------------------------------------
>> recv 411 bytes from udp/[203.208.207.212]:5060 at 03:10:30.824765:
>>   ------------------------------------------------------------------------
>>   SIP/2.0 487 Request Terminated
>>   Via: SIP/2.0/UDP
>> 10.1.1.46:5080;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>   To:
>> &lt;sip:03996563750914 at 203.208.207.212
>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>   CSeq: 13501649 INVITE
>>   Reason: SIP;cause=487;text="Request Terminated"
>>   Content-Length:     0
>>
>>   ------------------------------------------------------------------------
>> send 371 bytes to udp/[203.208.207.212]:5060 at 03:10:30.824891:
>>   ------------------------------------------------------------------------
>>   ACK sip:03996563750914 at 203.208.207.212 SIP/2.0
>>   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKZ49rj5Qtr8aBN
>>   Max-Forwards: 70
>>   From: "" &lt;sip:03996590031055 at 202.73.56.46&gt;;tag=jDSe6ycN62Zar
>>   To:
>> &lt;sip:03996563750914 at 203.208.207.212
>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>   CSeq: 13501649 ACK
>>   Content-Length: 0
>>
>>   ------------------------------------------------------------------------
>> recv 380 bytes from udp/[203.208.207.212]:5060 at 03:10:30.826375:
>>   ------------------------------------------------------------------------
>>   SIP/2.0 200 Ok
>>   Via: SIP/2.0/UDP
>> 10.1.1.46:5080;rport=5080;branch=z9hG4bK0D3Hm08XNH1Xg;received=10.1.1.46
>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>   To:
>> &lt;sip:03996590031055 at 203.208.207.212
>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>   CSeq: 13501634 BYE
>>
>>
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/Leg-A-is-automatically-disconnected-on-Leg-B-orginate-failure-tp6482192p6482192.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Fri Jun 17 09:57:35 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 17 Jun 2011 07:57:35 +0200
Subject: [Freeswitch-users] Setting Up an Sip Trunk without
	Authentication
In-Reply-To: <271B8C7B3AE54D7CA9229E49D3AB652D@clancysystems.com>
References: <75EE58D2AC7E4BFE9A53B699375DD699@clancysystems.com><BANLkTinwXNF4zyONuvBOpiDmrXwqYiv=Yw@mail.gmail.com><5A9A76E6D31143F1B2C971E2A7508D22@clancysystems.com><BANLkTimg6hEv0Vbg_+WkjKcuADp5dz0hNA@mail.gmail.com><AF706099B82D4F309EDBEA0ACE7DA5D4@clancysystems.com><BANLkTimR8-x4yWk=QFymZymVVK2JdPt_-g@mail.gmail.com><0580E32182F04516A074B86F364EDB5E@clancysystems.com>
	<BANLkTi=+aniSP1qUpVejYc-hksk51V8wMA@mail.gmail.com>
	<271B8C7B3AE54D7CA9229E49D3AB652D@clancysystems.com>
Message-ID: <3FC1A4D1-87F2-45AB-9AC8-F9C767F267B5@ipeva.fr>

Dave,

if you use the default config, I would recommend you configure Vitelity to send you the packets to your port 5080 (profile external).
Auth is disabled on external, so I think this port is more adapted to be used for unauth trunks.

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

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

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




Le 16/06/2011 ? 23:46, Dave a ?crit :

> Will Do. Thanks very much for your help.  I bought the book:)
>  
> Dave
>  
> ----- Original Message -----
> From: Michael Collins
> To: FreeSWITCH Users Help
> Sent: Thursday, June 16, 2011 3:22 PM
> Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication
> 
> Okay, I want you to go read up on gather data and dropping it on pastebin. The skills you hone there will server you well. :)
> 
> http://wiki.freeswitch.org/wiki/Reporting_Bugs
> 
> Need a debug log and a sip trace. Drop it on pastebin. Hint: use "sofia global siptrace on" to get all sip traffic logged to the console.
> 
> Put the pb link in this email thread.
> -MC
> 
> On Thu, Jun 16, 2011 at 2:18 PM, Dave <dave at clancysystems.com> wrote:
> I did both, and even restarted the service. same results. Velocity did a call capture on thier end and sent it to me. They say that FreeSWITCH is requiring proxy authentication and that that is the issue.
> SIP Status: 407 Proxy Authentication Required
>  
> ----- Original Message -----
> From: Michael Collins
> To: FreeSWITCH Users Help
> Sent: Thursday, June 16, 2011 2:46 PM
> Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication
> 
> Did you reloadxml and restart the sofia profile after the changes? Also, you need to "reloadacl" after doing a change to acl.conf.xml.
> 
> -MC
> 
> On Thu, Jun 16, 2011 at 1:27 PM, Dave <dave at clancysystems.com> wrote:
> Yes, I am only testing inbound.  When I call from our office phone (not connected to FreeSWITCH, it's on a trunk from broadvox)  I Get a busy signal and at that moment FreeSWITCH shows...
> "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone at theirIP] from their IP.
>  
> If I call from my cell phone I get the Operator Message "your call did not go through" and at that moment the same thing shows in FreeSWITCH
> "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone at theirIP] from their IP.
>  
>  
> ----- Original Message -----
> From: Michael Collins
> To: FreeSWITCH Users Help
> Sent: Thursday, June 16, 2011 2:11 PM
> Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication
> 
> 
> 
> On Thu, Jun 16, 2011 at 1:02 PM, Dave <dave at clancysystems.com> wrote:
>  
> Under "domains" in acl.conf.xml I added  
>  
>    <node type="allow" value="xxx.xxx.xxx.xxx"/>
>  
> The correct syntax is:
> <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>
>  
> x's are the IP address it's just a single address not a CIDR.
>  
>  
> And I created the file velocity_did.xml with the following code..
>  
> <include>
>   <extension name="velocity_did">
> 
>     <condition field="destination_number" expression="^(2064001950)$">
>       <action application="set" data="domain_name=$${domain}"/>
>       <action application="answer"/>
> 
>    <action application="lua" data="cme_ivr.lua channel_name"/>
>  
>  </condition>
>   </extension>
> </include>
> I still just get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone at theirIP] from their IP, but nothing more.
>  
> Wait - can you explain *exactly* what you're doing for testing? Are you trying to dial out via Velocity and back in to your DID? If so then you have 2 completely different things you need to set up. The instructions I gave were only for inbound DID, so use a cell phone or something to test that.
> 
> For outbound it seems like they are sending you an auth challenge, which means they need to give you a username and password as well as a host/ip. You need to create a gateway, preferably in your external profile. Just be sure to set the "register" param to false since they are not expecting you to register with them. 
> 
> -MC
> 
> Do I need to change the line ..
>  
> <action application="set" data="domain_name=$${domain}"/>
> to the IP address? If so what syntax?
>  
> They have no Domain Name.
>  
> Thanks again for your help.
>  
> Dave Goodwin     
>  
> ----- Original Message -----
> From: Michael Collins
> To: FreeSWITCH Users Help
> Sent: Thursday, June 16, 2011 12:15 PM
> Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication
> 
> Dave,
> 
> If you are simply handling incoming calls then you will need to do two things:
> 
> Add Velocity's IP addr to the domains section of acl.conf.xml
> Add an extension to the public context (conf/dialplan/public.xml)
> 
> Allowing a call into FS via an ACL will send it into the "public" context; from there you need to transfer it to the default context or just send it straight to your Lua script.
> 
> Also, I don't believe you need to use a "bridge" app based on the description you gave. Bridge is used to create a new outbound call leg (B leg) and connect it to the inbound call leg (A leg). If you just are handling a call with an IVR then there is no B leg needed, thus no need to bridge.
> 
> Welcome to FreeSWITCH, btw! :) I can highly recommend the FS book (the "bridge" book, ironically) as it discusses some of these basic concepts that will make your FS experience a whole lot more pleasant.
> 
> -MC
> 
> On Thu, Jun 16, 2011 at 8:54 AM, Dave <dave at clancysystems.com> wrote:
> Hi,
>  
> I am new to FreeSWITCH and need to set up FreeSWITCH to receive calls through a Velocity Networks SIP trunk. They say I should not authenticate with Username and Password. Rather, I connect directly to the their IP address.
>  
> I put
>  
> <extension name="velocity">
>       <condition field="destination_number" expression="^2064001950$">
>         <action application="bridge" data="sofia/external/$ {destination_number}@their_ip_address"/>
>         <action application="lua" data="cme_ivr.lua channel_name"/>
>      </condition>
> </extension>
>  
> in the default.xml under Dialplan
>  
> When I Dial the DID from a Phone (not one connected to the FreeSWITCH server) I get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone at theirIP] from their IP, but nothing more.
>  
> What I hope to do is take the incoming call from 2064001950 and route it to the Lua IVR script above.
>  
> I Appreciate any help you may offer.
>  
> Dave Goodwin
>  
>  
>  
>  
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From gcd at i.ph  Fri Jun 17 10:07:54 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Fri, 17 Jun 2011 14:07:54 +0800
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <BANLkTi=5LJDDxzY2d3nirUnQGpOqoBC45Q@mail.gmail.com>
References: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
	<201106162131.04859.justlikeef@gmail.com>
	<BANLkTi=5LJDDxzY2d3nirUnQGpOqoBC45Q@mail.gmail.com>
Message-ID: <BANLkTimY5oG_1-Tpk5eBKKC1xX8fXk3AoQ@mail.gmail.com>

hello guys!

i'm trying to setup a small exchange w/ Internet service - of course, using
FS as the exchange. DSLAMs connect to POTS via 2-wire splitters and that
would require using FXS gateway (as what Rob mentioned).

i'm trying to find out if there's a way we can omit the FXS - so it's
FS-to-DSLAM direct via IP.  i'm not sure if this setup is now possible on
newer IP DSLAMs.


On Fri, Jun 17, 2011 at 10:45 AM, jay binks <jaybinks at gmail.com> wrote:

> this is MY take on what Nandy is after.
>
> A Dslam provides DSL on certain frequencies of the line..
> ( some of ) the other frequencies are used for voice.
>
> my understanding is that sometimes this is split off to another device to
> provide the voice,
> but in this case Nandy is after a DSlam that will do the DSL part AND the
> voice frequencies ( Voice signalling / audio by SIP / RTP )
>
> I guess this is a logical question and would be quite interesting to see if
> there are such devices..
> sorry I dont know of any :(
>
> Jay
>
>
> On Fri, Jun 17, 2011 at 11:31 AM, Rob Hutton <justlikeef at gmail.com> wrote:
>
>> Can you explain further what you are trying to do?  A DSLAM is a device
>> that provides DSL transport, which is independent of what you run across it.
>>  So normally, you have
>>
>> |Server | ----|network|----|DSLAM| --------[dsl line] -------[modem or
>> modem/router combo]-----[network]---[Client]
>>
>> What the Server is (Could be Freeswitch) and what the Client is (could be
>> a VOIP phone) are independent of the transport???  Onewire, Cisco, and
>> others make some DSL Modem/Router combos with integrated Voip to FXS ports
>> for the CPE end.
>>
>> If you find a DSLAM running one of the OSs that Freeswitch supports, you
>> could compile Freeswitch for it, but I haven't seen one with enough CPU tp
>> handle much besides the traffic.
>>
>> On Thursday 16 June 2011 19:03:50 Nandy Dagondon wrote:
>> > hello everybody,
>> >
>> > i'm looking for small (24/48 ports)  IP DSLAMs that inter-connects
>> directly
>> > to FreeSwitch via IP.  is this already available on the market? or do we
>> > still have to connect FreeSwitch via POTS splitters and FXS gateways? i
>> > appreciate if you can mention some brands/models.
>> >
>> > tks,
>> > nandy
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely
>
> Jay
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From oseslija at gmail.com  Fri Jun 17 10:08:25 2011
From: oseslija at gmail.com (Ognjen Seslija)
Date: Fri, 17 Jun 2011 08:08:25 +0200
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <201106162131.04859.justlikeef@gmail.com>
References: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
	<201106162131.04859.justlikeef@gmail.com>
Message-ID: <BANLkTim9t098TTw7Ryynxy+8QyWtDTNdyA@mail.gmail.com>

I think he wants the newer DSLAMs without ATM layer.
On Jun 17, 2011 3:35 AM, "Rob Hutton" <justlikeef at gmail.com> wrote:
> Can you explain further what you are trying to do? A DSLAM is a device
that provides DSL transport, which is independent of what you run across it.
So normally, you have
>
> |Server | ----|network|----|DSLAM| --------[dsl line] -------[modem or
modem/router combo]-----[network]---[Client]
>
> What the Server is (Could be Freeswitch) and what the Client is (could be
a VOIP phone) are independent of the transport??? Onewire, Cisco, and others
make some DSL Modem/Router combos with integrated Voip to FXS ports for the
CPE end.
>
> If you find a DSLAM running one of the OSs that Freeswitch supports, you
could compile Freeswitch for it, but I haven't seen one with enough CPU tp
handle much besides the traffic.
>
> On Thursday 16 June 2011 19:03:50 Nandy Dagondon wrote:
>> hello everybody,
>>
>> i'm looking for small (24/48 ports) IP DSLAMs that inter-connects
directly
>> to FreeSwitch via IP. is this already available on the market? or do we
>> still have to connect FreeSwitch via POTS splitters and FXS gateways? i
>> appreciate if you can mention some brands/models.
>>
>> tks,
>> nandy
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From david.ponzone at ipeva.fr  Fri Jun 17 10:19:12 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 17 Jun 2011 08:19:12 +0200
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <BANLkTimY5oG_1-Tpk5eBKKC1xX8fXk3AoQ@mail.gmail.com>
References: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
	<201106162131.04859.justlikeef@gmail.com>
	<BANLkTi=5LJDDxzY2d3nirUnQGpOqoBC45Q@mail.gmail.com>
	<BANLkTimY5oG_1-Tpk5eBKKC1xX8fXk3AoQ@mail.gmail.com>
Message-ID: <C201E991-C3A1-437E-BC22-C82FD7360FC1@ipeva.fr>

Which density are you looking for ?
Is it for use in a private building or on public lines ?

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

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

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




Le 17/06/2011 ? 08:07, Nandy Dagondon a ?crit :

> hello guys!
> 
> i'm trying to setup a small exchange w/ Internet service - of course, using FS as the exchange. DSLAMs connect to POTS via 2-wire splitters and that would require using FXS gateway (as what Rob mentioned).  
> 
> i'm trying to find out if there's a way we can omit the FXS - so it's FS-to-DSLAM direct via IP.  i'm not sure if this setup is now possible on newer IP DSLAMs. 
> 
> 
> On Fri, Jun 17, 2011 at 10:45 AM, jay binks <jaybinks at gmail.com> wrote:
> this is MY take on what Nandy is after.
> 
> A Dslam provides DSL on certain frequencies of the line..
> ( some of ) the other frequencies are used for voice.
> 
> my understanding is that sometimes this is split off to another device to provide the voice,
> but in this case Nandy is after a DSlam that will do the DSL part AND the voice frequencies ( Voice signalling / audio by SIP / RTP )
> 
> I guess this is a logical question and would be quite interesting to see if there are such devices..
> sorry I dont know of any :(
> 
> Jay
> 
> 
> On Fri, Jun 17, 2011 at 11:31 AM, Rob Hutton <justlikeef at gmail.com> wrote:
> Can you explain further what you are trying to do?  A DSLAM is a device that provides DSL transport, which is independent of what you run across it.  So normally, you have
> 
> |Server | ----|network|----|DSLAM| --------[dsl line] -------[modem or modem/router combo]-----[network]---[Client]
> 
> What the Server is (Could be Freeswitch) and what the Client is (could be a VOIP phone) are independent of the transport???  Onewire, Cisco, and others make some DSL Modem/Router combos with integrated Voip to FXS ports for the CPE end.
> 
> If you find a DSLAM running one of the OSs that Freeswitch supports, you could compile Freeswitch for it, but I haven't seen one with enough CPU tp handle much besides the traffic.
> 
> On Thursday 16 June 2011 19:03:50 Nandy Dagondon wrote:
> > hello everybody,
> >
> > i'm looking for small (24/48 ports)  IP DSLAMs that inter-connects directly
> > to FreeSwitch via IP.  is this already available on the market? or do we
> > still have to connect FreeSwitch via POTS splitters and FXS gateways? i
> > appreciate if you can mention some brands/models.
> >
> > tks,
> > nandy
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> -- 
> Sincerely
> 
> Jay
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From gcd at i.ph  Fri Jun 17 10:28:54 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Fri, 17 Jun 2011 14:28:54 +0800
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <C201E991-C3A1-437E-BC22-C82FD7360FC1@ipeva.fr>
References: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
	<201106162131.04859.justlikeef@gmail.com>
	<BANLkTi=5LJDDxzY2d3nirUnQGpOqoBC45Q@mail.gmail.com>
	<BANLkTimY5oG_1-Tpk5eBKKC1xX8fXk3AoQ@mail.gmail.com>
	<C201E991-C3A1-437E-BC22-C82FD7360FC1@ipeva.fr>
Message-ID: <BANLkTin5YSNPPkQfLNuht=yTxmgKeRWAVw@mail.gmail.com>

yes, without ATM layer and it's for public lines.



On Fri, Jun 17, 2011 at 2:19 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> Which density are you looking for ?
> Is it for use in a private building or on public lines ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 17/06/2011 ? 08:07, Nandy Dagondon a ?crit :
>
> hello guys!
>
> i'm trying to setup a small exchange w/ Internet service - of course, using
> FS as the exchange. DSLAMs connect to POTS via 2-wire splitters and that
> would require using FXS gateway (as what Rob mentioned).
>
> i'm trying to find out if there's a way we can omit the FXS - so it's
> FS-to-DSLAM direct via IP.  i'm not sure if this setup is now possible on
> newer IP DSLAMs.
>
>
> On Fri, Jun 17, 2011 at 10:45 AM, jay binks <jaybinks at gmail.com> wrote:
>
>> this is MY take on what Nandy is after.
>>
>> A Dslam provides DSL on certain frequencies of the line..
>> ( some of ) the other frequencies are used for voice.
>>
>> my understanding is that sometimes this is split off to another device to
>> provide the voice,
>> but in this case Nandy is after a DSlam that will do the DSL part AND the
>> voice frequencies ( Voice signalling / audio by SIP / RTP )
>>
>> I guess this is a logical question and would be quite interesting to see
>> if there are such devices..
>> sorry I dont know of any :(
>>
>> Jay
>>
>>
>> On Fri, Jun 17, 2011 at 11:31 AM, Rob Hutton <justlikeef at gmail.com>wrote:
>>
>>> Can you explain further what you are trying to do?  A DSLAM is a device
>>> that provides DSL transport, which is independent of what you run across it.
>>>  So normally, you have
>>>
>>> |Server | ----|network|----|DSLAM| --------[dsl line] -------[modem or
>>> modem/router combo]-----[network]---[Client]
>>>
>>> What the Server is (Could be Freeswitch) and what the Client is (could be
>>> a VOIP phone) are independent of the transport???  Onewire, Cisco, and
>>> others make some DSL Modem/Router combos with integrated Voip to FXS ports
>>> for the CPE end.
>>>
>>> If you find a DSLAM running one of the OSs that Freeswitch supports, you
>>> could compile Freeswitch for it, but I haven't seen one with enough CPU tp
>>> handle much besides the traffic.
>>>
>>> On Thursday 16 June 2011 19:03:50 Nandy Dagondon wrote:
>>> > hello everybody,
>>> >
>>> > i'm looking for small (24/48 ports)  IP DSLAMs that inter-connects
>>> directly
>>> > to FreeSwitch via IP.  is this already available on the market? or do
>>> we
>>> > still have to connect FreeSwitch via POTS splitters and FXS gateways? i
>>> > appreciate if you can mention some brands/models.
>>> >
>>> > tks,
>>> > nandy
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Sincerely
>>
>> Jay
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gcd at i.ph  Fri Jun 17 10:30:06 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Fri, 17 Jun 2011 14:30:06 +0800
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <BANLkTin5YSNPPkQfLNuht=yTxmgKeRWAVw@mail.gmail.com>
References: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
	<201106162131.04859.justlikeef@gmail.com>
	<BANLkTi=5LJDDxzY2d3nirUnQGpOqoBC45Q@mail.gmail.com>
	<BANLkTimY5oG_1-Tpk5eBKKC1xX8fXk3AoQ@mail.gmail.com>
	<C201E991-C3A1-437E-BC22-C82FD7360FC1@ipeva.fr>
	<BANLkTin5YSNPPkQfLNuht=yTxmgKeRWAVw@mail.gmail.com>
Message-ID: <BANLkTi=C8Qnsjvr8utxf5xOaR74f69H_aQ@mail.gmail.com>

density - 24 or 48 lines only



On Fri, Jun 17, 2011 at 2:28 PM, Nandy Dagondon <gcd at i.ph> wrote:

> yes, without ATM layer and it's for public lines.
>
>
>
>
> On Fri, Jun 17, 2011 at 2:19 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>
>> Which density are you looking for ?
>> Is it for use in a private building or on public lines ?
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 17/06/2011 ? 08:07, Nandy Dagondon a ?crit :
>>
>> hello guys!
>>
>> i'm trying to setup a small exchange w/ Internet service - of course,
>> using FS as the exchange. DSLAMs connect to POTS via 2-wire splitters and
>> that would require using FXS gateway (as what Rob mentioned).
>>
>> i'm trying to find out if there's a way we can omit the FXS - so it's
>> FS-to-DSLAM direct via IP.  i'm not sure if this setup is now possible on
>> newer IP DSLAMs.
>>
>>
>> On Fri, Jun 17, 2011 at 10:45 AM, jay binks <jaybinks at gmail.com> wrote:
>>
>>> this is MY take on what Nandy is after.
>>>
>>> A Dslam provides DSL on certain frequencies of the line..
>>> ( some of ) the other frequencies are used for voice.
>>>
>>> my understanding is that sometimes this is split off to another device to
>>> provide the voice,
>>> but in this case Nandy is after a DSlam that will do the DSL part AND the
>>> voice frequencies ( Voice signalling / audio by SIP / RTP )
>>>
>>> I guess this is a logical question and would be quite interesting to see
>>> if there are such devices..
>>> sorry I dont know of any :(
>>>
>>> Jay
>>>
>>>
>>> On Fri, Jun 17, 2011 at 11:31 AM, Rob Hutton <justlikeef at gmail.com>wrote:
>>>
>>>> Can you explain further what you are trying to do?  A DSLAM is a device
>>>> that provides DSL transport, which is independent of what you run across it.
>>>>  So normally, you have
>>>>
>>>> |Server | ----|network|----|DSLAM| --------[dsl line] -------[modem or
>>>> modem/router combo]-----[network]---[Client]
>>>>
>>>> What the Server is (Could be Freeswitch) and what the Client is (could
>>>> be a VOIP phone) are independent of the transport???  Onewire, Cisco, and
>>>> others make some DSL Modem/Router combos with integrated Voip to FXS ports
>>>> for the CPE end.
>>>>
>>>> If you find a DSLAM running one of the OSs that Freeswitch supports, you
>>>> could compile Freeswitch for it, but I haven't seen one with enough CPU tp
>>>> handle much besides the traffic.
>>>>
>>>> On Thursday 16 June 2011 19:03:50 Nandy Dagondon wrote:
>>>> > hello everybody,
>>>> >
>>>> > i'm looking for small (24/48 ports)  IP DSLAMs that inter-connects
>>>> directly
>>>> > to FreeSwitch via IP.  is this already available on the market? or do
>>>> we
>>>> > still have to connect FreeSwitch via POTS splitters and FXS gateways?
>>>> i
>>>> > appreciate if you can mention some brands/models.
>>>> >
>>>> > tks,
>>>> > nandy
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Sincerely
>>>
>>> Jay
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From Stefan.Weigel at allianz-warranty.com  Fri Jun 17 11:10:49 2011
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Fri, 17 Jun 2011 09:10:49 +0200
Subject: [Freeswitch-users] mod_spandsp event spandsp::rxfaxresult
 variables missing
In-Reply-To: <BANLkTikLeff9RL=8mOMvdOjUf998n1aL-A@mail.gmail.com>
References: <5003D7D3E06F514E8C682F18D223265C04D3B36C2B@AZWSMS03.azwarranty.int>
	<BANLkTin3vQii2arQd4KhmSBDoJFivaLiBQ@mail.gmail.com>
	<5003D7D3E06F514E8C682F18D223265C04D3B36C2F@AZWSMS03.azwarranty.int>
	<BANLkTikLeff9RL=8mOMvdOjUf998n1aL-A@mail.gmail.com>
Message-ID: <5003D7D3E06F514E8C682F18D223265C04D3B36C30@AZWSMS03.azwarranty.int>

Hi Michael,



I'm lazy and so I'm using the freeswitch-snapshot.tar.gz from files.freeswitch.org.

Should I always do a 'make current' to get the latest version ?





Best regards



Stefan





Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Collins
Gesendet: Donnerstag, 16. Juni 2011 20:18
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] mod_spandsp event spandsp::rxfaxresult variables missing





On Thu, Jun 16, 2011 at 3:25 AM, Weigel, Stefan <Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>> wrote:

Hi Steve,



version shows me only 'git-.



This usually means that you have a git version older than 1.7.x and should probably get a new one. I think we're at least at git 1.7.4...



-MC

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From ankitwalia4u at gmail.com  Fri Jun 17 11:25:32 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Fri, 17 Jun 2011 12:55:32 +0530
Subject: [Freeswitch-users] Making a external call to Mobile number
In-Reply-To: <BANLkTinaMATA-tCcTDi367JFK9a9YG4OAw@mail.gmail.com>
References: <BANLkTinFuhpjGkLv14UC3LGUYcFc8H9cjQ@mail.gmail.com>
	<BANLkTinaMATA-tCcTDi367JFK9a9YG4OAw@mail.gmail.com>
Message-ID: <BANLkTin8pmecsF86EbAXsjdNo5m-O2JJPw@mail.gmail.com>

Yes, the sofia status is REGED.

I am adding my Dial Plan and SIP Profile

SIP Profile

<include>
  <gateway name="actiovoip">
    <param name="username" value="XXXX"/>
    <param name="password" value="XXXX"/>
    <param name="realm" value="actionvoip.com"/>
    <param name="proxy" value="sip.actionvoip.com"/>
  </gateway>
</include>

-----------------------------------------------------------
DialPlan

<include>
  <extension name="Dial Out Actionvoip Gateway">
    <condition field="destination_number" expression="^9(91\d{10})$">
      <action application="bridge" data="sofia/gateway/actionvoip/$1"/>
    </condition>
  </extension>
</include>

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

In the logs, I found Errors. Full logs at Pastebin
http://pastebin.freeswitch.org/16541

2011-06-17 12:45:14.794565 [ERR] mod_sofia.c:3183 Invalid Gateway

2011-06-17 12:45:14.794565 [ERR] switch_ivr_originate.c:2430 Cannot create
outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]

I am stuck now. I did try most of the permutations and combinations.

Thanks

Ankit




On Fri, Jun 17, 2011 at 4:08 AM, Nandy Dagondon <gcd at i.ph> wrote:

> in fs_cli, execute *sofia status* to check if the account is REGED. the
> account should be in  sip_profiles/external/. show us a sample of the
> account entry and the dialplan.
>
>
> On Fri, Jun 17, 2011 at 1:04 AM, ankIT WALiA <ankitwalia4u at gmail.com>wrote:
>
>> Dear all,
>>
>> I am trying to make an external call to a mobile number of India using the
>> default ext number 1005. For doing this, I did the following things.
>>
>> First, I created a user at IPTEL.org, I added a SIP profile. I registered
>> with IPTEL. I also added a dial plan regex for 91 - (10 digit number). But,
>> because of some issue, I was not able to connect to IPTEL SIP. I was getting
>> error on X-LITE about some server issue.
>>
>> then, I tried my VOIP account with ActionVoip which is a paid account and
>> has SIP service. I could connect to my action voip account through X-LITE
>> SIP.
>>
>> I made a SIP profile for Action Voip and registered the username with
>> action voip in FS. On reload, it got registered using action voip SIP
>> profile. I changed the dialplan to use actionvoip gateway.
>>
>> But still, when I am trying to call to my mobile number in India. The call
>> is not able to connect.
>>
>> I think there is gap in my knowledge. Please help me I am new to
>> FreeSwitch.
>>
>> Thanks
>> Ankit
>>
>> --
>> Life is like a rose its upto u feel it as its fragrance or thorns
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Life is like a rose its upto u feel it as its fragrance or thorns
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From steveayre at gmail.com  Fri Jun 17 11:31:15 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 17 Jun 2011 08:31:15 +0100
Subject: [Freeswitch-users] Making a external call to Mobile number
In-Reply-To: <BANLkTin8pmecsF86EbAXsjdNo5m-O2JJPw@mail.gmail.com>
References: <BANLkTinFuhpjGkLv14UC3LGUYcFc8H9cjQ@mail.gmail.com>
	<BANLkTinaMATA-tCcTDi367JFK9a9YG4OAw@mail.gmail.com>
	<BANLkTin8pmecsF86EbAXsjdNo5m-O2JJPw@mail.gmail.com>
Message-ID: <BANLkTi=8ohMsJwy0vwMF7KrkCiGcqF=jYQ@mail.gmail.com>

It's complaining the gateway you're trying to use doesn't exist. It looks
like a simple typo in the gateway name attribute:

<gateway name="*actiovoip*">
<action application="bridge" data="sofia/gateway/*actionvoip*/$1"/>

-Steve



On 17 June 2011 08:25, ankIT WALiA <ankitwalia4u at gmail.com> wrote:

> Yes, the sofia status is REGED.
>
> I am adding my Dial Plan and SIP Profile
>
> SIP Profile
>
> <include>
>   <gateway name="actiovoip">
>     <param name="username" value="XXXX"/>
>     <param name="password" value="XXXX"/>
>     <param name="realm" value="actionvoip.com"/>
>     <param name="proxy" value="sip.actionvoip.com"/>
>   </gateway>
> </include>
>
> -----------------------------------------------------------
> DialPlan
>
> <include>
>   <extension name="Dial Out Actionvoip Gateway">
>     <condition field="destination_number" expression="^9(91\d{10})$">
>       <action application="bridge" data="sofia/gateway/actionvoip/$1"/>
>     </condition>
>   </extension>
> </include>
>
> -----------------------------------------------------------
>
> In the logs, I found Errors. Full logs at Pastebin
> http://pastebin.freeswitch.org/16541
>
> 2011-06-17 12:45:14.794565 [ERR] mod_sofia.c:3183 Invalid Gateway
>
> 2011-06-17 12:45:14.794565 [ERR] switch_ivr_originate.c:2430 Cannot create
> outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
>
> I am stuck now. I did try most of the permutations and combinations.
>
> Thanks
>
> Ankit
>
>
>
>
> On Fri, Jun 17, 2011 at 4:08 AM, Nandy Dagondon <gcd at i.ph> wrote:
>
>> in fs_cli, execute *sofia status* to check if the account is REGED. the
>> account should be in  sip_profiles/external/. show us a sample of the
>> account entry and the dialplan.
>>
>>
>> On Fri, Jun 17, 2011 at 1:04 AM, ankIT WALiA <ankitwalia4u at gmail.com>wrote:
>>
>>> Dear all,
>>>
>>> I am trying to make an external call to a mobile number of India using
>>> the default ext number 1005. For doing this, I did the following things.
>>>
>>> First, I created a user at IPTEL.org, I added a SIP profile. I registered
>>> with IPTEL. I also added a dial plan regex for 91 - (10 digit number). But,
>>> because of some issue, I was not able to connect to IPTEL SIP. I was getting
>>> error on X-LITE about some server issue.
>>>
>>> then, I tried my VOIP account with ActionVoip which is a paid account and
>>> has SIP service. I could connect to my action voip account through X-LITE
>>> SIP.
>>>
>>> I made a SIP profile for Action Voip and registered the username with
>>> action voip in FS. On reload, it got registered using action voip SIP
>>> profile. I changed the dialplan to use actionvoip gateway.
>>>
>>> But still, when I am trying to call to my mobile number in India. The
>>> call is not able to connect.
>>>
>>> I think there is gap in my knowledge. Please help me I am new to
>>> FreeSwitch.
>>>
>>> Thanks
>>> Ankit
>>>
>>> --
>>> Life is like a rose its upto u feel it as its fragrance or thorns
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Life is like a rose its upto u feel it as its fragrance or thorns
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Fri Jun 17 11:32:54 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 17 Jun 2011 08:32:54 +0100
Subject: [Freeswitch-users] mod_spandsp event spandsp::rxfaxresult
 variables missing
In-Reply-To: <5003D7D3E06F514E8C682F18D223265C04D3B36C30@AZWSMS03.azwarranty.int>
References: <5003D7D3E06F514E8C682F18D223265C04D3B36C2B@AZWSMS03.azwarranty.int>
	<BANLkTin3vQii2arQd4KhmSBDoJFivaLiBQ@mail.gmail.com>
	<5003D7D3E06F514E8C682F18D223265C04D3B36C2F@AZWSMS03.azwarranty.int>
	<BANLkTikLeff9RL=8mOMvdOjUf998n1aL-A@mail.gmail.com>
	<5003D7D3E06F514E8C682F18D223265C04D3B36C30@AZWSMS03.azwarranty.int>
Message-ID: <BANLkTi=zJUTLM4-5Y6nUys3MRpLWRR4eSg@mail.gmail.com>

Check it out using Git:
'git clone git://git.freeswitch.org/freeswitch.git'

-Steve


On 17 June 2011 08:10, Weigel, Stefan <Stefan.Weigel at allianz-warranty.com>wrote:

>  Hi Michael,
>
>
>
> I?m lazy and so I?m using the freeswitch-snapshot.tar.gz from
> files.freeswitch.org.
>
> Should I always do a ?make current? to get the latest version ?
>
>
>
>
>
> Best regards
>
>
>
> Stefan
>
>
>
>
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Michael
> Collins
> *Gesendet:* Donnerstag, 16. Juni 2011 20:18
>
> *An:* FreeSWITCH Users Help
> *Betreff:* Re: [Freeswitch-users] mod_spandsp event spandsp::rxfaxresult
> variables missing
>
>
>
>
>
> On Thu, Jun 16, 2011 at 3:25 AM, Weigel, Stefan <
> Stefan.Weigel at allianz-warranty.com> wrote:
>
> Hi Steve,
>
>
>
> version shows me only ?git-.
>
>
>
> This usually means that you have a git version older than 1.7.x and should
> probably get a new one. I think we're at least at git 1.7.4...
>
>
>
> -MC
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Fri Jun 17 11:37:34 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 17 Jun 2011 09:37:34 +0200
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <BANLkTin5YSNPPkQfLNuht=yTxmgKeRWAVw@mail.gmail.com>
References: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
	<201106162131.04859.justlikeef@gmail.com>
	<BANLkTi=5LJDDxzY2d3nirUnQGpOqoBC45Q@mail.gmail.com>
	<BANLkTimY5oG_1-Tpk5eBKKC1xX8fXk3AoQ@mail.gmail.com>
	<C201E991-C3A1-437E-BC22-C82FD7360FC1@ipeva.fr>
	<BANLkTin5YSNPPkQfLNuht=yTxmgKeRWAVw@mail.gmail.com>
Message-ID: <45FF3438-C239-48D9-A57B-F0A49166900E@ipeva.fr>

Nandy,

something does not add up.
When you deploy DSLAMs for public lines, it's not for 24/48 ports, but rather for 20 000 to 30 000 ports.
Perhaps there is a confusion about what is a public line.
A public line is the 200 meters-8km pair of copper going from the local (legacy) telco facility to your house.
You don't install your own DSLAMs for 24/48 ports, or perhaps you are in a very specific situation.
I really doubt you will find a cost effective DSLAM for so few ports.
Plus, you realize you will need to install on DSLAM in all the telco facilities required to get the coverage you need

There are small DSLAMs on the market, but they are targeted for private lines (hospitality, health care, ...) and I really don't know if they can work on public lines.

In case you actually need a small DSLAM just to run ADSL over a private pair of copper, you should have a look at:

http://www.phybridge.com/uniphyer-ip-phones.aspx

This thing does not call itself a DSLAM but it is one.
They market that as a "switch" that enables LAN on your copper wires.
It's a small lie for data usage as you only get 25Mbps/1.4Mbps (ATM), but for voice, that's enough.
Basically, you put the "switch"-DSLAM in the network cabinet where the copper wires go, you connect the wires to it, you connect the ethernet port to your LAN (so your FS). In each room/office, you plug an adapter (provided also by Phybridge) which is basically a small ADSL modem (powered from the switch through the wire), from which you get an ethernet port, with POE on it!

I like the solution on the paper, but never used. Because of the cost:
a 24 ports switch is 3500$
one adapter is 122$

so a full 24 ports solution would be 6500$, which is not very competitive compared to recabling the place.

But your mileage may vary.

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

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

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




Le 17/06/2011 ? 08:28, Nandy Dagondon a ?crit :

> yes, without ATM layer and it's for public lines.
> 
> 
> 
> On Fri, Jun 17, 2011 at 2:19 PM, David Ponzone <david.ponzone at ipeva.fr> wrote:
> Which density are you looking for ?
> Is it for use in a private building or on public lines ?
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 17/06/2011 ? 08:07, Nandy Dagondon a ?crit :
> 
>> hello guys!
>> 
>> i'm trying to setup a small exchange w/ Internet service - of course, using FS as the exchange. DSLAMs connect to POTS via 2-wire splitters and that would require using FXS gateway (as what Rob mentioned).  
>> 
>> i'm trying to find out if there's a way we can omit the FXS - so it's FS-to-DSLAM direct via IP.  i'm not sure if this setup is now possible on newer IP DSLAMs. 
>> 
>> 
>> On Fri, Jun 17, 2011 at 10:45 AM, jay binks <jaybinks at gmail.com> wrote:
>> this is MY take on what Nandy is after.
>> 
>> A Dslam provides DSL on certain frequencies of the line..
>> ( some of ) the other frequencies are used for voice.
>> 
>> my understanding is that sometimes this is split off to another device to provide the voice,
>> but in this case Nandy is after a DSlam that will do the DSL part AND the voice frequencies ( Voice signalling / audio by SIP / RTP )
>> 
>> I guess this is a logical question and would be quite interesting to see if there are such devices..
>> sorry I dont know of any :(
>> 
>> Jay
>> 
>> 
>> On Fri, Jun 17, 2011 at 11:31 AM, Rob Hutton <justlikeef at gmail.com> wrote:
>> Can you explain further what you are trying to do?  A DSLAM is a device that provides DSL transport, which is independent of what you run across it.  So normally, you have
>> 
>> |Server | ----|network|----|DSLAM| --------[dsl line] -------[modem or modem/router combo]-----[network]---[Client]
>> 
>> What the Server is (Could be Freeswitch) and what the Client is (could be a VOIP phone) are independent of the transport???  Onewire, Cisco, and others make some DSL Modem/Router combos with integrated Voip to FXS ports for the CPE end.
>> 
>> If you find a DSLAM running one of the OSs that Freeswitch supports, you could compile Freeswitch for it, but I haven't seen one with enough CPU tp handle much besides the traffic.
>> 
>> On Thursday 16 June 2011 19:03:50 Nandy Dagondon wrote:
>> > hello everybody,
>> >
>> > i'm looking for small (24/48 ports)  IP DSLAMs that inter-connects directly
>> > to FreeSwitch via IP.  is this already available on the market? or do we
>> > still have to connect FreeSwitch via POTS splitters and FXS gateways? i
>> > appreciate if you can mention some brands/models.
>> >
>> > tks,
>> > nandy
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> -- 
>> Sincerely
>> 
>> Jay
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From ankitwalia4u at gmail.com  Fri Jun 17 11:51:31 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Fri, 17 Jun 2011 13:21:31 +0530
Subject: [Freeswitch-users] Making a external call to Mobile number
In-Reply-To: <BANLkTi=8ohMsJwy0vwMF7KrkCiGcqF=jYQ@mail.gmail.com>
References: <BANLkTinFuhpjGkLv14UC3LGUYcFc8H9cjQ@mail.gmail.com>
	<BANLkTinaMATA-tCcTDi367JFK9a9YG4OAw@mail.gmail.com>
	<BANLkTin8pmecsF86EbAXsjdNo5m-O2JJPw@mail.gmail.com>
	<BANLkTi=8ohMsJwy0vwMF7KrkCiGcqF=jYQ@mail.gmail.com>
Message-ID: <BANLkTinythv8JFtwML9Uv8yd5j7FGz8BOw@mail.gmail.com>

Ahhh! I wasted a half day in figuring out what's the issue?
Thanks a lot Steve.

On Fri, Jun 17, 2011 at 1:01 PM, Steven Ayre <steveayre at gmail.com> wrote:

> It's complaining the gateway you're trying to use doesn't exist. It looks
> like a simple typo in the gateway name attribute:
>
> <gateway name="*actiovoip*">
>
> <action application="bridge" data="sofia/gateway/*actionvoip*/$1"/>
>
> -Steve
>
>
>
>
> On 17 June 2011 08:25, ankIT WALiA <ankitwalia4u at gmail.com> wrote:
>
>> Yes, the sofia status is REGED.
>>
>> I am adding my Dial Plan and SIP Profile
>>
>> SIP Profile
>>
>> <include>
>>   <gateway name="actiovoip">
>>     <param name="username" value="XXXX"/>
>>     <param name="password" value="XXXX"/>
>>     <param name="realm" value="actionvoip.com"/>
>>     <param name="proxy" value="sip.actionvoip.com"/>
>>   </gateway>
>> </include>
>>
>> -----------------------------------------------------------
>> DialPlan
>>
>> <include>
>>   <extension name="Dial Out Actionvoip Gateway">
>>     <condition field="destination_number" expression="^9(91\d{10})$">
>>       <action application="bridge" data="sofia/gateway/actionvoip/$1"/>
>>     </condition>
>>   </extension>
>> </include>
>>
>> -----------------------------------------------------------
>>
>> In the logs, I found Errors. Full logs at Pastebin
>> http://pastebin.freeswitch.org/16541
>>
>> 2011-06-17 12:45:14.794565 [ERR] mod_sofia.c:3183 Invalid Gateway
>>
>> 2011-06-17 12:45:14.794565 [ERR] switch_ivr_originate.c:2430 Cannot create
>> outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
>>
>> I am stuck now. I did try most of the permutations and combinations.
>>
>> Thanks
>>
>> Ankit
>>
>>
>>
>>
>> On Fri, Jun 17, 2011 at 4:08 AM, Nandy Dagondon <gcd at i.ph> wrote:
>>
>>> in fs_cli, execute *sofia status* to check if the account is REGED. the
>>> account should be in  sip_profiles/external/. show us a sample of the
>>> account entry and the dialplan.
>>>
>>>
>>> On Fri, Jun 17, 2011 at 1:04 AM, ankIT WALiA <ankitwalia4u at gmail.com>wrote:
>>>
>>>> Dear all,
>>>>
>>>> I am trying to make an external call to a mobile number of India using
>>>> the default ext number 1005. For doing this, I did the following things.
>>>>
>>>> First, I created a user at IPTEL.org, I added a SIP profile. I
>>>> registered with IPTEL. I also added a dial plan regex for 91 - (10 digit
>>>> number). But, because of some issue, I was not able to connect to IPTEL SIP.
>>>> I was getting error on X-LITE about some server issue.
>>>>
>>>> then, I tried my VOIP account with ActionVoip which is a paid account
>>>> and has SIP service. I could connect to my action voip account through
>>>> X-LITE SIP.
>>>>
>>>> I made a SIP profile for Action Voip and registered the username with
>>>> action voip in FS. On reload, it got registered using action voip SIP
>>>> profile. I changed the dialplan to use actionvoip gateway.
>>>>
>>>> But still, when I am trying to call to my mobile number in India. The
>>>> call is not able to connect.
>>>>
>>>> I think there is gap in my knowledge. Please help me I am new to
>>>> FreeSwitch.
>>>>
>>>> Thanks
>>>> Ankit
>>>>
>>>> --
>>>> Life is like a rose its upto u feel it as its fragrance or thorns
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Life is like a rose its upto u feel it as its fragrance or thorns
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Life is like a rose its upto u feel it as its fragrance or thorns
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From brent at overthewire.com.au  Fri Jun 17 12:10:29 2011
From: brent at overthewire.com.au (Brent Paddon)
Date: Fri, 17 Jun 2011 18:10:29 +1000
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <BANLkTin5YSNPPkQfLNuht=yTxmgKeRWAVw@mail.gmail.com>
References: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
	<201106162131.04859.justlikeef@gmail.com>
	<BANLkTi=5LJDDxzY2d3nirUnQGpOqoBC45Q@mail.gmail.com>
	<BANLkTimY5oG_1-Tpk5eBKKC1xX8fXk3AoQ@mail.gmail.com>
	<C201E991-C3A1-437E-BC22-C82FD7360FC1@ipeva.fr>
	<BANLkTin5YSNPPkQfLNuht=yTxmgKeRWAVw@mail.gmail.com>
Message-ID: <BANLkTi=XiZ21pS5NDaPLiSip+P6+VfKRSQ@mail.gmail.com>

I presume you are actually after an MSAN :

http://en.wikipedia.org/wiki/Multi-service_access_node

An MSAN allows the single copper pair to be used for voice and data.  The
voice component appears to the subscriber as a standard POTS and the MSAN
then uses SIP to connect back to a softswitch such as FS.

Perhaps something like this:

http://www.zhone.com/products/msan/MALC-XP/

Brent

On Fri, Jun 17, 2011 at 4:28 PM, Nandy Dagondon <gcd at i.ph> wrote:

> yes, without ATM layer and it's for public lines.
>
>
>
>
> On Fri, Jun 17, 2011 at 2:19 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>
>> Which density are you looking for ?
>> Is it for use in a private building or on public lines ?
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 17/06/2011 ? 08:07, Nandy Dagondon a ?crit :
>>
>> hello guys!
>>
>> i'm trying to setup a small exchange w/ Internet service - of course,
>> using FS as the exchange. DSLAMs connect to POTS via 2-wire splitters and
>> that would require using FXS gateway (as what Rob mentioned).
>>
>> i'm trying to find out if there's a way we can omit the FXS - so it's
>> FS-to-DSLAM direct via IP.  i'm not sure if this setup is now possible on
>> newer IP DSLAMs.
>>
>>
>> On Fri, Jun 17, 2011 at 10:45 AM, jay binks <jaybinks at gmail.com> wrote:
>>
>>> this is MY take on what Nandy is after.
>>>
>>> A Dslam provides DSL on certain frequencies of the line..
>>> ( some of ) the other frequencies are used for voice.
>>>
>>> my understanding is that sometimes this is split off to another device to
>>> provide the voice,
>>> but in this case Nandy is after a DSlam that will do the DSL part AND the
>>> voice frequencies ( Voice signalling / audio by SIP / RTP )
>>>
>>> I guess this is a logical question and would be quite interesting to see
>>> if there are such devices..
>>> sorry I dont know of any :(
>>>
>>> Jay
>>>
>>>
>>> On Fri, Jun 17, 2011 at 11:31 AM, Rob Hutton <justlikeef at gmail.com>wrote:
>>>
>>>> Can you explain further what you are trying to do?  A DSLAM is a device
>>>> that provides DSL transport, which is independent of what you run across it.
>>>>  So normally, you have
>>>>
>>>> |Server | ----|network|----|DSLAM| --------[dsl line] -------[modem or
>>>> modem/router combo]-----[network]---[Client]
>>>>
>>>> What the Server is (Could be Freeswitch) and what the Client is (could
>>>> be a VOIP phone) are independent of the transport???  Onewire, Cisco, and
>>>> others make some DSL Modem/Router combos with integrated Voip to FXS ports
>>>> for the CPE end.
>>>>
>>>> If you find a DSLAM running one of the OSs that Freeswitch supports, you
>>>> could compile Freeswitch for it, but I haven't seen one with enough CPU tp
>>>> handle much besides the traffic.
>>>>
>>>> On Thursday 16 June 2011 19:03:50 Nandy Dagondon wrote:
>>>> > hello everybody,
>>>> >
>>>> > i'm looking for small (24/48 ports)  IP DSLAMs that inter-connects
>>>> directly
>>>> > to FreeSwitch via IP.  is this already available on the market? or do
>>>> we
>>>> > still have to connect FreeSwitch via POTS splitters and FXS gateways?
>>>> i
>>>> > appreciate if you can mention some brands/models.
>>>> >
>>>> > tks,
>>>> > nandy
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Sincerely
>>>
>>> Jay
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
--
Brent Paddon

Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
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From gcd at i.ph  Fri Jun 17 12:25:59 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Fri, 17 Jun 2011 16:25:59 +0800
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <45FF3438-C239-48D9-A57B-F0A49166900E@ipeva.fr>
References: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
	<201106162131.04859.justlikeef@gmail.com>
	<BANLkTi=5LJDDxzY2d3nirUnQGpOqoBC45Q@mail.gmail.com>
	<BANLkTimY5oG_1-Tpk5eBKKC1xX8fXk3AoQ@mail.gmail.com>
	<C201E991-C3A1-437E-BC22-C82FD7360FC1@ipeva.fr>
	<BANLkTin5YSNPPkQfLNuht=yTxmgKeRWAVw@mail.gmail.com>
	<45FF3438-C239-48D9-A57B-F0A49166900E@ipeva.fr>
Message-ID: <BANLkTin3apQVJLXh-novL0deaq+QM0TuTQ@mail.gmail.com>

yes, it doesn't add up because the application is in a rural area. the
Phybridge is a good solution except for the price. thanks David. however,
the MSAN, perfectly describes what i'm looking - hopefully the investment is
lower.

thanks to all who contributed.

On Fri, Jun 17, 2011 at 3:37 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> Nandy,
>
> something does not add up.
> When you deploy DSLAMs for public lines, it's not for 24/48 ports, but
> rather for 20 000 to 30 000 ports.
> Perhaps there is a confusion about what is a public line.
> A public line is the 200 meters-8km pair of copper going from the local
> (legacy) telco facility to your house.
> You don't install your own DSLAMs for 24/48 ports, or perhaps you are in a
> very specific situation.
> I really doubt you will find a cost effective DSLAM for so few ports.
> Plus, you realize you will need to install on DSLAM in all the telco
> facilities required to get the coverage you need
>
> There are small DSLAMs on the market, but they are targeted for private
> lines (hospitality, health care, ...) and I really don't know if they can
> work on public lines.
>
> In case you actually need a small DSLAM just to run ADSL over a private
> pair of copper, you should have a look at:
>
> http://www.phybridge.com/uniphyer-ip-phones.aspx
>
> This thing does not call itself a DSLAM but it is one.
> They market that as a "switch" that enables LAN on your copper wires.
> It's a small lie for data usage as you only get 25Mbps/1.4Mbps (ATM), but
> for voice, that's enough.
> Basically, you put the "switch"-DSLAM in the network cabinet where the
> copper wires go, you connect the wires to it, you connect the ethernet port
> to your LAN (so your FS). In each room/office, you plug an adapter (provided
> also by Phybridge) which is basically a small ADSL modem (powered from the
> switch through the wire), from which you get an ethernet port, with POE on
> it!
>
> I like the solution on the paper, but never used. Because of the cost:
> a 24 ports switch is 3500$
> one adapter is 122$
>
> so a full 24 ports solution would be 6500$, which is not very competitive
> compared to recabling the place.
>
> But your mileage may vary.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 17/06/2011 ? 08:28, Nandy Dagondon a ?crit :
>
> yes, without ATM layer and it's for public lines.
>
>
>
> On Fri, Jun 17, 2011 at 2:19 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>
>> Which density are you looking for ?
>> Is it for use in a private building or on public lines ?
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 17/06/2011 ? 08:07, Nandy Dagondon a ?crit :
>>
>> hello guys!
>>
>> i'm trying to setup a small exchange w/ Internet service - of course,
>> using FS as the exchange. DSLAMs connect to POTS via 2-wire splitters and
>> that would require using FXS gateway (as what Rob mentioned).
>>
>> i'm trying to find out if there's a way we can omit the FXS - so it's
>> FS-to-DSLAM direct via IP.  i'm not sure if this setup is now possible on
>> newer IP DSLAMs.
>>
>>
>> On Fri, Jun 17, 2011 at 10:45 AM, jay binks <jaybinks at gmail.com> wrote:
>>
>>> this is MY take on what Nandy is after.
>>>
>>> A Dslam provides DSL on certain frequencies of the line..
>>> ( some of ) the other frequencies are used for voice.
>>>
>>> my understanding is that sometimes this is split off to another device to
>>> provide the voice,
>>> but in this case Nandy is after a DSlam that will do the DSL part AND the
>>> voice frequencies ( Voice signalling / audio by SIP / RTP )
>>>
>>> I guess this is a logical question and would be quite interesting to see
>>> if there are such devices..
>>> sorry I dont know of any :(
>>>
>>> Jay
>>>
>>>
>>> On Fri, Jun 17, 2011 at 11:31 AM, Rob Hutton <justlikeef at gmail.com>wrote:
>>>
>>>> Can you explain further what you are trying to do?  A DSLAM is a device
>>>> that provides DSL transport, which is independent of what you run across it.
>>>>  So normally, you have
>>>>
>>>> |Server | ----|network|----|DSLAM| --------[dsl line] -------[modem or
>>>> modem/router combo]-----[network]---[Client]
>>>>
>>>> What the Server is (Could be Freeswitch) and what the Client is (could
>>>> be a VOIP phone) are independent of the transport???  Onewire, Cisco, and
>>>> others make some DSL Modem/Router combos with integrated Voip to FXS ports
>>>> for the CPE end.
>>>>
>>>> If you find a DSLAM running one of the OSs that Freeswitch supports, you
>>>> could compile Freeswitch for it, but I haven't seen one with enough CPU tp
>>>> handle much besides the traffic.
>>>>
>>>> On Thursday 16 June 2011 19:03:50 Nandy Dagondon wrote:
>>>> > hello everybody,
>>>> >
>>>> > i'm looking for small (24/48 ports)  IP DSLAMs that inter-connects
>>>> directly
>>>> > to FreeSwitch via IP.  is this already available on the market? or do
>>>> we
>>>> > still have to connect FreeSwitch via POTS splitters and FXS gateways?
>>>> i
>>>> > appreciate if you can mention some brands/models.
>>>> >
>>>> > tks,
>>>> > nandy
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Sincerely
>>>
>>> Jay
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Fri Jun 17 13:29:48 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 17 Jun 2011 11:29:48 +0200
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <BANLkTin3apQVJLXh-novL0deaq+QM0TuTQ@mail.gmail.com>
References: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
	<201106162131.04859.justlikeef@gmail.com>
	<BANLkTi=5LJDDxzY2d3nirUnQGpOqoBC45Q@mail.gmail.com>
	<BANLkTimY5oG_1-Tpk5eBKKC1xX8fXk3AoQ@mail.gmail.com>
	<C201E991-C3A1-437E-BC22-C82FD7360FC1@ipeva.fr>
	<BANLkTin5YSNPPkQfLNuht=yTxmgKeRWAVw@mail.gmail.com>
	<45FF3438-C239-48D9-A57B-F0A49166900E@ipeva.fr>
	<BANLkTin3apQVJLXh-novL0deaq+QM0TuTQ@mail.gmail.com>
Message-ID: <0013F8A2-8D82-470C-B8D8-45C22BDB3A00@ipeva.fr>

Hmmm unlikely I think.
The Zhone box for instance seems to be around 11,000$.

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

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

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




Le 17/06/2011 ? 10:25, Nandy Dagondon a ?crit :

> yes, it doesn't add up because the application is in a rural area. the Phybridge is a good solution except for the price. thanks David. however, the MSAN, perfectly describes what i'm looking - hopefully the investment is lower.
> 
> thanks to all who contributed.
>  
> On Fri, Jun 17, 2011 at 3:37 PM, David Ponzone <david.ponzone at ipeva.fr> wrote:
> Nandy,
> 
> something does not add up.
> When you deploy DSLAMs for public lines, it's not for 24/48 ports, but rather for 20 000 to 30 000 ports.
> Perhaps there is a confusion about what is a public line.
> A public line is the 200 meters-8km pair of copper going from the local (legacy) telco facility to your house.
> You don't install your own DSLAMs for 24/48 ports, or perhaps you are in a very specific situation.
> I really doubt you will find a cost effective DSLAM for so few ports.
> Plus, you realize you will need to install on DSLAM in all the telco facilities required to get the coverage you need
> 
> There are small DSLAMs on the market, but they are targeted for private lines (hospitality, health care, ...) and I really don't know if they can work on public lines.
> 
> In case you actually need a small DSLAM just to run ADSL over a private pair of copper, you should have a look at:
> 
> http://www.phybridge.com/uniphyer-ip-phones.aspx
> 
> This thing does not call itself a DSLAM but it is one.
> They market that as a "switch" that enables LAN on your copper wires.
> It's a small lie for data usage as you only get 25Mbps/1.4Mbps (ATM), but for voice, that's enough.
> Basically, you put the "switch"-DSLAM in the network cabinet where the copper wires go, you connect the wires to it, you connect the ethernet port to your LAN (so your FS). In each room/office, you plug an adapter (provided also by Phybridge) which is basically a small ADSL modem (powered from the switch through the wire), from which you get an ethernet port, with POE on it!
> 
> I like the solution on the paper, but never used. Because of the cost:
> a 24 ports switch is 3500$
> one adapter is 122$
> 
> so a full 24 ports solution would be 6500$, which is not very competitive compared to recabling the place.
> 
> But your mileage may vary.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 17/06/2011 ? 08:28, Nandy Dagondon a ?crit :
> 
>> yes, without ATM layer and it's for public lines.
>> 
>> 
>> 
>> On Fri, Jun 17, 2011 at 2:19 PM, David Ponzone <david.ponzone at ipeva.fr> wrote:
>> Which density are you looking for ?
>> Is it for use in a private building or on public lines ?
>> 
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> 
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> 
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>> 
>> 
>> 
>> 
>> Le 17/06/2011 ? 08:07, Nandy Dagondon a ?crit :
>> 
>>> hello guys!
>>> 
>>> i'm trying to setup a small exchange w/ Internet service - of course, using FS as the exchange. DSLAMs connect to POTS via 2-wire splitters and that would require using FXS gateway (as what Rob mentioned).  
>>> 
>>> i'm trying to find out if there's a way we can omit the FXS - so it's FS-to-DSLAM direct via IP.  i'm not sure if this setup is now possible on newer IP DSLAMs. 
>>> 
>>> 
>>> On Fri, Jun 17, 2011 at 10:45 AM, jay binks <jaybinks at gmail.com> wrote:
>>> this is MY take on what Nandy is after.
>>> 
>>> A Dslam provides DSL on certain frequencies of the line..
>>> ( some of ) the other frequencies are used for voice.
>>> 
>>> my understanding is that sometimes this is split off to another device to provide the voice,
>>> but in this case Nandy is after a DSlam that will do the DSL part AND the voice frequencies ( Voice signalling / audio by SIP / RTP )
>>> 
>>> I guess this is a logical question and would be quite interesting to see if there are such devices..
>>> sorry I dont know of any :(
>>> 
>>> Jay
>>> 
>>> 
>>> On Fri, Jun 17, 2011 at 11:31 AM, Rob Hutton <justlikeef at gmail.com> wrote:
>>> Can you explain further what you are trying to do?  A DSLAM is a device that provides DSL transport, which is independent of what you run across it.  So normally, you have
>>> 
>>> |Server | ----|network|----|DSLAM| --------[dsl line] -------[modem or modem/router combo]-----[network]---[Client]
>>> 
>>> What the Server is (Could be Freeswitch) and what the Client is (could be a VOIP phone) are independent of the transport???  Onewire, Cisco, and others make some DSL Modem/Router combos with integrated Voip to FXS ports for the CPE end.
>>> 
>>> If you find a DSLAM running one of the OSs that Freeswitch supports, you could compile Freeswitch for it, but I haven't seen one with enough CPU tp handle much besides the traffic.
>>> 
>>> On Thursday 16 June 2011 19:03:50 Nandy Dagondon wrote:
>>> > hello everybody,
>>> >
>>> > i'm looking for small (24/48 ports)  IP DSLAMs that inter-connects directly
>>> > to FreeSwitch via IP.  is this already available on the market? or do we
>>> > still have to connect FreeSwitch via POTS splitters and FXS gateways? i
>>> > appreciate if you can mention some brands/models.
>>> >
>>> > tks,
>>> > nandy
>>> 
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> 
>>> -- 
>>> Sincerely
>>> 
>>> Jay
>>> 
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From gcd at i.ph  Fri Jun 17 14:57:44 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Fri, 17 Jun 2011 18:57:44 +0800
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <0013F8A2-8D82-470C-B8D8-45C22BDB3A00@ipeva.fr>
References: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
	<201106162131.04859.justlikeef@gmail.com>
	<BANLkTi=5LJDDxzY2d3nirUnQGpOqoBC45Q@mail.gmail.com>
	<BANLkTimY5oG_1-Tpk5eBKKC1xX8fXk3AoQ@mail.gmail.com>
	<C201E991-C3A1-437E-BC22-C82FD7360FC1@ipeva.fr>
	<BANLkTin5YSNPPkQfLNuht=yTxmgKeRWAVw@mail.gmail.com>
	<45FF3438-C239-48D9-A57B-F0A49166900E@ipeva.fr>
	<BANLkTin3apQVJLXh-novL0deaq+QM0TuTQ@mail.gmail.com>
	<0013F8A2-8D82-470C-B8D8-45C22BDB3A00@ipeva.fr>
Message-ID: <BANLkTinLho7z+XgBHGjUJY7+vyaWV60NMA@mail.gmail.com>

huh ... that's quiet pricey. perhaps an IP DSLAM + FXS would be a cheaper
setup. in rural areas, many are still dinosaurs.


On Fri, Jun 17, 2011 at 5:29 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> Hmmm unlikely I think.
> The Zhone box for instance seems to be around 11,000$.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 17/06/2011 ? 10:25, Nandy Dagondon a ?crit :
>
> yes, it doesn't add up because the application is in a rural area. the
> Phybridge is a good solution except for the price. thanks David. however,
> the MSAN, perfectly describes what i'm looking - hopefully the investment is
> lower.
>
> thanks to all who contributed.
>
> On Fri, Jun 17, 2011 at 3:37 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>
>> Nandy,
>>
>> something does not add up.
>> When you deploy DSLAMs for public lines, it's not for 24/48 ports, but
>> rather for 20 000 to 30 000 ports.
>> Perhaps there is a confusion about what is a public line.
>> A public line is the 200 meters-8km pair of copper going from the local
>> (legacy) telco facility to your house.
>> You don't install your own DSLAMs for 24/48 ports, or perhaps you are in a
>> very specific situation.
>> I really doubt you will find a cost effective DSLAM for so few ports.
>> Plus, you realize you will need to install on DSLAM in all the telco
>> facilities required to get the coverage you need
>>
>> There are small DSLAMs on the market, but they are targeted for private
>> lines (hospitality, health care, ...) and I really don't know if they can
>> work on public lines.
>>
>> In case you actually need a small DSLAM just to run ADSL over a private
>> pair of copper, you should have a look at:
>>
>> http://www.phybridge.com/uniphyer-ip-phones.aspx
>>
>> This thing does not call itself a DSLAM but it is one.
>> They market that as a "switch" that enables LAN on your copper wires.
>> It's a small lie for data usage as you only get 25Mbps/1.4Mbps (ATM), but
>> for voice, that's enough.
>> Basically, you put the "switch"-DSLAM in the network cabinet where the
>> copper wires go, you connect the wires to it, you connect the ethernet port
>> to your LAN (so your FS). In each room/office, you plug an adapter (provided
>> also by Phybridge) which is basically a small ADSL modem (powered from the
>> switch through the wire), from which you get an ethernet port, with POE on
>> it!
>>
>> I like the solution on the paper, but never used. Because of the cost:
>> a 24 ports switch is 3500$
>> one adapter is 122$
>>
>> so a full 24 ports solution would be 6500$, which is not very competitive
>> compared to recabling the place.
>>
>> But your mileage may vary.
>>
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 17/06/2011 ? 08:28, Nandy Dagondon a ?crit :
>>
>> yes, without ATM layer and it's for public lines.
>>
>>
>>
>> On Fri, Jun 17, 2011 at 2:19 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>>
>>> Which density are you looking for ?
>>> Is it for use in a private building or on public lines ?
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 17/06/2011 ? 08:07, Nandy Dagondon a ?crit :
>>>
>>> hello guys!
>>>
>>> i'm trying to setup a small exchange w/ Internet service - of course,
>>> using FS as the exchange. DSLAMs connect to POTS via 2-wire splitters and
>>> that would require using FXS gateway (as what Rob mentioned).
>>>
>>> i'm trying to find out if there's a way we can omit the FXS - so it's
>>> FS-to-DSLAM direct via IP.  i'm not sure if this setup is now possible on
>>> newer IP DSLAMs.
>>>
>>>
>>> On Fri, Jun 17, 2011 at 10:45 AM, jay binks <jaybinks at gmail.com> wrote:
>>>
>>>> this is MY take on what Nandy is after.
>>>>
>>>> A Dslam provides DSL on certain frequencies of the line..
>>>> ( some of ) the other frequencies are used for voice.
>>>>
>>>> my understanding is that sometimes this is split off to another device
>>>> to provide the voice,
>>>> but in this case Nandy is after a DSlam that will do the DSL part AND
>>>> the voice frequencies ( Voice signalling / audio by SIP / RTP )
>>>>
>>>> I guess this is a logical question and would be quite interesting to see
>>>> if there are such devices..
>>>> sorry I dont know of any :(
>>>>
>>>> Jay
>>>>
>>>>
>>>> On Fri, Jun 17, 2011 at 11:31 AM, Rob Hutton <justlikeef at gmail.com>wrote:
>>>>
>>>>> Can you explain further what you are trying to do?  A DSLAM is a device
>>>>> that provides DSL transport, which is independent of what you run across it.
>>>>>  So normally, you have
>>>>>
>>>>> |Server | ----|network|----|DSLAM| --------[dsl line] -------[modem or
>>>>> modem/router combo]-----[network]---[Client]
>>>>>
>>>>> What the Server is (Could be Freeswitch) and what the Client is (could
>>>>> be a VOIP phone) are independent of the transport???  Onewire, Cisco, and
>>>>> others make some DSL Modem/Router combos with integrated Voip to FXS ports
>>>>> for the CPE end.
>>>>>
>>>>> If you find a DSLAM running one of the OSs that Freeswitch supports,
>>>>> you could compile Freeswitch for it, but I haven't seen one with enough CPU
>>>>> tp handle much besides the traffic.
>>>>>
>>>>> On Thursday 16 June 2011 19:03:50 Nandy Dagondon wrote:
>>>>> > hello everybody,
>>>>> >
>>>>> > i'm looking for small (24/48 ports)  IP DSLAMs that inter-connects
>>>>> directly
>>>>> > to FreeSwitch via IP.  is this already available on the market? or do
>>>>> we
>>>>> > still have to connect FreeSwitch via POTS splitters and FXS gateways?
>>>>> i
>>>>> > appreciate if you can mention some brands/models.
>>>>> >
>>>>> > tks,
>>>>> > nandy
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Sincerely
>>>>
>>>> Jay
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From justlikeef at gmail.com  Fri Jun 17 14:57:58 2011
From: justlikeef at gmail.com (Rob Hutton)
Date: Fri, 17 Jun 2011 06:57:58 -0400
Subject: [Freeswitch-users] Unable to get TLS working
Message-ID: <201106170657.58978.justlikeef@gmail.com>

I have been unable to get TLS working for several days despite beating my head 
against it, so I am sure I am doing something simple and stupid.  Brian and 
Jeff have tried to help, but I haven't gotten anywhere.  I would appreciate 
any help that I can get....

http://jira.freeswitch.org/browse/FS-3346

Thanks,
Rob


From mbrancaleoni at voismart.it  Fri Jun 17 15:14:30 2011
From: mbrancaleoni at voismart.it (Matteo)
Date: Fri, 17 Jun 2011 13:14:30 +0200 (CEST)
Subject: [Freeswitch-users] Generating ringback if 180 arrives after 183
Message-ID: <1f3159c1-33ac-42f1-8277-fcb0b79829c7@mx.voismart.com>

Hi all,

I've the following scenario from my upstream provider :

FS calls provider
Provider sends easrly media with 183 announcing that the call is going to be trasferred
Provider sends 180 without sdp

In my dialplan I'm using ringback variable to let FS generate ringback
to internal network, and if I receive 180 without 183 is working ok.

What I'm trying to achieve is to let ringback work even if we've already received
the 183 with sdp.

I've tryed a lot of dialplan magic without luck.
Maybe we need to add this support into FS itself ?

What I see seems that we need to break the bridge or at least
"inject" the tone stream into the internal leg of the call.

To be fair, the exact sequence is:

FS calls provider
provider sends 183/sdp annoucing the call trasnfer
provider sends update which stops the media (media=inactive)
provider sends 180 ringing (we must generate here)

maybe the correct way is to handle the update, put fs back
into ringing state and proceed from there?
Is feasible or we can get the same result with some dialplan magic?

any hint is appreciated :)

regards,
matteo


From spencer at 5ninesolutions.com  Fri Jun 17 15:26:29 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Fri, 17 Jun 2011 04:26:29 -0700
Subject: [Freeswitch-users] Kamailio Interop
Message-ID: <A8C633C2-9FAD-4221-AEAB-95BDEF79CFA2@5ninesolutions.com>

Hello all,
I've been working on upgrading our SBC setup which consists of FreeSWITCH instances between Kamailio 3.1 instances.  In initial testing FreeSWITCH would fail on an ACK timeout.  After looking at the traces, Kamailio was responding to the 200 with an ACK that has a branch=0 parameter.  It is possible to disable this behavior by setting syn_branch=0 in kamailio.cfg which causes an MD5 hash to be used as the branch parameter.  In researching this it seems that this has been around in SER for some time as the ACK following the 200 is sent statelessly regardless of a t_relay() function.


See:
http://www.mail-archive.com/users at openser.org/msg00697.html

The question I have is there any way to get Freeswitch to work with an ACK with a branch=0?  and if so should it?

Gotta love SIP.  :-)

Spencer



From spencer at 5ninesolutions.com  Fri Jun 17 15:39:47 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Fri, 17 Jun 2011 04:39:47 -0700
Subject: [Freeswitch-users] Kamailio Interop
In-Reply-To: <A8C633C2-9FAD-4221-AEAB-95BDEF79CFA2@5ninesolutions.com>
References: <A8C633C2-9FAD-4221-AEAB-95BDEF79CFA2@5ninesolutions.com>
Message-ID: <B1801B32-22CB-48E2-8E98-C4AB66EA9228@5ninesolutions.com>

And since this a Kamailio issue, I don't want to dwell on it but, here's the fix if anyone else runs into this problem...

http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commitdiff;h=ebb3b085c15b398192cd8e242d46914252278448


On Jun 17, 2011, at 4:26 AM, Spencer Thomason wrote:

> Hello all,
> I've been working on upgrading our SBC setup which consists of FreeSWITCH instances between Kamailio 3.1 instances.  In initial testing FreeSWITCH would fail on an ACK timeout.  After looking at the traces, Kamailio was responding to the 200 with an ACK that has a branch=0 parameter.  It is possible to disable this behavior by setting syn_branch=0 in kamailio.cfg which causes an MD5 hash to be used as the branch parameter.  In researching this it seems that this has been around in SER for some time as the ACK following the 200 is sent statelessly regardless of a t_relay() function.
> 
> 
> See:
> http://www.mail-archive.com/users at openser.org/msg00697.html
> 
> The question I have is there any way to get Freeswitch to work with an ACK with a branch=0?  and if so should it?
> 
> Gotta love SIP.  :-)
> 
> Spencer
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From brent at overthewire.com.au  Fri Jun 17 15:54:37 2011
From: brent at overthewire.com.au (Brent Paddon)
Date: Fri, 17 Jun 2011 21:54:37 +1000
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <BANLkTinLho7z+XgBHGjUJY7+vyaWV60NMA@mail.gmail.com>
References: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
	<201106162131.04859.justlikeef@gmail.com>
	<BANLkTi=5LJDDxzY2d3nirUnQGpOqoBC45Q@mail.gmail.com>
	<BANLkTimY5oG_1-Tpk5eBKKC1xX8fXk3AoQ@mail.gmail.com>
	<C201E991-C3A1-437E-BC22-C82FD7360FC1@ipeva.fr>
	<BANLkTin5YSNPPkQfLNuht=yTxmgKeRWAVw@mail.gmail.com>
	<45FF3438-C239-48D9-A57B-F0A49166900E@ipeva.fr>
	<BANLkTin3apQVJLXh-novL0deaq+QM0TuTQ@mail.gmail.com>
	<0013F8A2-8D82-470C-B8D8-45C22BDB3A00@ipeva.fr>
	<BANLkTinLho7z+XgBHGjUJY7+vyaWV60NMA@mail.gmail.com>
Message-ID: <BANLkTin32MjC2Q8K9VOZj27wQUbwv4osFg@mail.gmail.com>

Probably just check what distance your FXS solution will support.  I've
heard a number of them do not support long distances commonly found on
outdoor style plant (they're more intended for in-building).

Brent

On Fri, Jun 17, 2011 at 8:57 PM, Nandy Dagondon <gcd at i.ph> wrote:

> huh ... that's quiet pricey. perhaps an IP DSLAM + FXS would be a cheaper
> setup. in rural areas, many are still dinosaurs.
>
>
>
> On Fri, Jun 17, 2011 at 5:29 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>
>> Hmmm unlikely I think.
>> The Zhone box for instance seems to be around 11,000$.
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 17/06/2011 ? 10:25, Nandy Dagondon a ?crit :
>>
>> yes, it doesn't add up because the application is in a rural area. the
>> Phybridge is a good solution except for the price. thanks David. however,
>> the MSAN, perfectly describes what i'm looking - hopefully the investment is
>> lower.
>>
>> thanks to all who contributed.
>>
>> On Fri, Jun 17, 2011 at 3:37 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>>
>>> Nandy,
>>>
>>> something does not add up.
>>> When you deploy DSLAMs for public lines, it's not for 24/48 ports, but
>>> rather for 20 000 to 30 000 ports.
>>> Perhaps there is a confusion about what is a public line.
>>> A public line is the 200 meters-8km pair of copper going from the local
>>> (legacy) telco facility to your house.
>>> You don't install your own DSLAMs for 24/48 ports, or perhaps you are in
>>> a very specific situation.
>>> I really doubt you will find a cost effective DSLAM for so few ports.
>>> Plus, you realize you will need to install on DSLAM in all the telco
>>> facilities required to get the coverage you need
>>>
>>> There are small DSLAMs on the market, but they are targeted for private
>>> lines (hospitality, health care, ...) and I really don't know if they can
>>> work on public lines.
>>>
>>> In case you actually need a small DSLAM just to run ADSL over a private
>>> pair of copper, you should have a look at:
>>>
>>> http://www.phybridge.com/uniphyer-ip-phones.aspx
>>>
>>> This thing does not call itself a DSLAM but it is one.
>>> They market that as a "switch" that enables LAN on your copper wires.
>>> It's a small lie for data usage as you only get 25Mbps/1.4Mbps (ATM), but
>>> for voice, that's enough.
>>> Basically, you put the "switch"-DSLAM in the network cabinet where the
>>> copper wires go, you connect the wires to it, you connect the ethernet port
>>> to your LAN (so your FS). In each room/office, you plug an adapter (provided
>>> also by Phybridge) which is basically a small ADSL modem (powered from the
>>> switch through the wire), from which you get an ethernet port, with POE on
>>> it!
>>>
>>> I like the solution on the paper, but never used. Because of the cost:
>>> a 24 ports switch is 3500$
>>> one adapter is 122$
>>>
>>> so a full 24 ports solution would be 6500$, which is not very competitive
>>> compared to recabling the place.
>>>
>>> But your mileage may vary.
>>>
>>> David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 17/06/2011 ? 08:28, Nandy Dagondon a ?crit :
>>>
>>> yes, without ATM layer and it's for public lines.
>>>
>>>
>>>
>>> On Fri, Jun 17, 2011 at 2:19 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>>>
>>>> Which density are you looking for ?
>>>> Is it for use in a private building or on public lines ?
>>>>
>>>>  David Ponzone  Direction Technique
>>>> email: david.ponzone at ipeva.fr
>>>> tel:      01 74 03 18 97
>>>> gsm:   06 66 98 76 34
>>>>
>>>> Service Client IPeva
>>>> tel:      0811 46 26 26
>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>
>>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>>> l'exp?diteur.*
>>>> *
>>>> *
>>>>
>>>>
>>>>
>>>> Le 17/06/2011 ? 08:07, Nandy Dagondon a ?crit :
>>>>
>>>> hello guys!
>>>>
>>>> i'm trying to setup a small exchange w/ Internet service - of course,
>>>> using FS as the exchange. DSLAMs connect to POTS via 2-wire splitters and
>>>> that would require using FXS gateway (as what Rob mentioned).
>>>>
>>>> i'm trying to find out if there's a way we can omit the FXS - so it's
>>>> FS-to-DSLAM direct via IP.  i'm not sure if this setup is now possible on
>>>> newer IP DSLAMs.
>>>>
>>>>
>>>> On Fri, Jun 17, 2011 at 10:45 AM, jay binks <jaybinks at gmail.com> wrote:
>>>>
>>>>> this is MY take on what Nandy is after.
>>>>>
>>>>> A Dslam provides DSL on certain frequencies of the line..
>>>>> ( some of ) the other frequencies are used for voice.
>>>>>
>>>>> my understanding is that sometimes this is split off to another device
>>>>> to provide the voice,
>>>>> but in this case Nandy is after a DSlam that will do the DSL part AND
>>>>> the voice frequencies ( Voice signalling / audio by SIP / RTP )
>>>>>
>>>>> I guess this is a logical question and would be quite interesting to
>>>>> see if there are such devices..
>>>>> sorry I dont know of any :(
>>>>>
>>>>> Jay
>>>>>
>>>>>
>>>>> On Fri, Jun 17, 2011 at 11:31 AM, Rob Hutton <justlikeef at gmail.com>wrote:
>>>>>
>>>>>> Can you explain further what you are trying to do?  A DSLAM is a
>>>>>> device that provides DSL transport, which is independent of what you run
>>>>>> across it.  So normally, you have
>>>>>>
>>>>>> |Server | ----|network|----|DSLAM| --------[dsl line] -------[modem or
>>>>>> modem/router combo]-----[network]---[Client]
>>>>>>
>>>>>> What the Server is (Could be Freeswitch) and what the Client is (could
>>>>>> be a VOIP phone) are independent of the transport???  Onewire, Cisco, and
>>>>>> others make some DSL Modem/Router combos with integrated Voip to FXS ports
>>>>>> for the CPE end.
>>>>>>
>>>>>> If you find a DSLAM running one of the OSs that Freeswitch supports,
>>>>>> you could compile Freeswitch for it, but I haven't seen one with enough CPU
>>>>>> tp handle much besides the traffic.
>>>>>>
>>>>>> On Thursday 16 June 2011 19:03:50 Nandy Dagondon wrote:
>>>>>> > hello everybody,
>>>>>> >
>>>>>> > i'm looking for small (24/48 ports)  IP DSLAMs that inter-connects
>>>>>> directly
>>>>>> > to FreeSwitch via IP.  is this already available on the market? or
>>>>>> do we
>>>>>> > still have to connect FreeSwitch via POTS splitters and FXS
>>>>>> gateways? i
>>>>>> > appreciate if you can mention some brands/models.
>>>>>> >
>>>>>> > tks,
>>>>>> > nandy
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Sincerely
>>>>>
>>>>> Jay
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
--
Brent Paddon

Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
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From ankitwalia4u at gmail.com  Fri Jun 17 16:13:04 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Fri, 17 Jun 2011 17:43:04 +0530
Subject: [Freeswitch-users] No Voice for default IVR ext 5000
Message-ID: <BANLkTi=xZ09f_9HNH6D=c8fpTM72qXbxJA@mail.gmail.com>

Dear all,

I have configured an gateway for external profile to receive and make calls.

I have added a file in public directory.
<include>
<extension name="iptel-inbound">
<condition field="destination_number"
expression="^(MY_IPTEL_USERNAME)$">
<action application="set" data="domain_name=$${domain}"/>
<action application="transfer" data="ExtNo XML default"/>
</condition>
</extension>
</include>

Now, while I am calling to my voip account, I am transferring the call. It
is getting transferred to the extension also.
But, If I use normal ext number, I am able to talk both ways. But when I use
default ext number 5000 for IVR menu. I could not listen to IVR menu.
Though, I can see in my logs that the audio file are being played one by
one, but I cant listen.

Do I need some extra configuration to listen to IVR?
There may be a gap in my knowledge. Please enlighten.

Thanks
Ankit
-- 
Life is like a rose its upto u feel it as its fragrance or thorns
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From ankitwalia4u at gmail.com  Fri Jun 17 16:13:43 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Fri, 17 Jun 2011 17:43:43 +0530
Subject: [Freeswitch-users] No Voice for default IVR ext 5000
In-Reply-To: <BANLkTi=xZ09f_9HNH6D=c8fpTM72qXbxJA@mail.gmail.com>
References: <BANLkTi=xZ09f_9HNH6D=c8fpTM72qXbxJA@mail.gmail.com>
Message-ID: <BANLkTi=8w+NCZid6JA23oo95Sj58UOPxAg@mail.gmail.com>

I am able to listen to Ext 5000 IVR menu, if I call from internal ext.

On Fri, Jun 17, 2011 at 5:43 PM, ankIT WALiA <ankitwalia4u at gmail.com> wrote:

> Dear all,
>
> I have configured an gateway for external profile to receive and make
> calls.
>
> I have added a file in public directory.
> <include>
> <extension name="iptel-inbound">
> <condition field="destination_number"
> expression="^(MY_IPTEL_USERNAME)$">
> <action application="set" data="domain_name=$${domain}"/>
> <action application="transfer" data="ExtNo XML default"/>
> </condition>
> </extension>
> </include>
>
> Now, while I am calling to my voip account, I am transferring the call. It
> is getting transferred to the extension also.
> But, If I use normal ext number, I am able to talk both ways. But when I
> use default ext number 5000 for IVR menu. I could not listen to IVR menu.
> Though, I can see in my logs that the audio file are being played one by
> one, but I cant listen.
>
> Do I need some extra configuration to listen to IVR?
> There may be a gap in my knowledge. Please enlighten.
>
> Thanks
> Ankit
> --
> Life is like a rose its upto u feel it as its fragrance or thorns
>



-- 
Life is like a rose its upto u feel it as its fragrance or thorns
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From mitch.johnson7 at gmail.com  Fri Jun 17 16:52:51 2011
From: mitch.johnson7 at gmail.com (mitch Johnson)
Date: Fri, 17 Jun 2011 08:52:51 -0400
Subject: [Freeswitch-users] Message Numbers
Message-ID: <BANLkTik7Z4Wfoc9Dtc8SvZj7Mf1nxkyOoQ@mail.gmail.com>

Can we have the messages seperated by their respective message numbers.  I
subscribe to quite a few lists and the FreeSWITCH list is the only one that
doesn't transpose the message number down into the messages.  It would make
searching for specific messages much easier.

Thanks,

Mitch
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From fs-list at communicatefreely.net  Fri Jun 17 17:27:49 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 17 Jun 2011 09:27:49 -0400
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
References: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
Message-ID: <4DFB5655.5010204@communicatefreely.net>

Take a look at Zhone technologies.

The MXK is a really nice DSLAM for medium to high densities.  They have 
line cards for ADSL2+, ADSL2+ w/ POTS (exactly what you are after), 
G.SHDSL, ADSL2+ with splitters, POTS only, and GPON.  That pretty much 
covers everything.

The POTS line cards are essentially FXS gateways.  The combo cards give 
you ADSL2+ and a SIP ATA essentially on the same board.  This all 
terminates out to Gigabit Ethernet, so you can connect it up to your IP 
network and use FreeSwitch to handle the voice side of things.  I think 
they even make a T1 card for that frame.

If you want something small, take a look at their Raptor DSLAMs.  These 
are 1U models that are environmentally hardened and can be used in 
outdoor cabinets.  I'm pretty sure they make a model that has 24 ports 
of ADSL2+ and POTS.

Good luck!

-Tim

Nandy Dagondon wrote:
> hello everybody,
>
> i'm looking for small (24/48 ports)  IP DSLAMs that inter-connects 
> directly to FreeSwitch via IP.  is this already available on the 
> market? or do we still have to connect FreeSwitch via POTS splitters 
> and FXS gateways? i appreciate if you can mention some brands/models.
>
> tks,
> nandy
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From steveayre at gmail.com  Fri Jun 17 17:35:52 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 17 Jun 2011 14:35:52 +0100
Subject: [Freeswitch-users] No Voice for default IVR ext 5000
In-Reply-To: <BANLkTi=xZ09f_9HNH6D=c8fpTM72qXbxJA@mail.gmail.com>
References: <BANLkTi=xZ09f_9HNH6D=c8fpTM72qXbxJA@mail.gmail.com>
Message-ID: <BANLkTinr17B8y2EcwQ0YAVwAGU3giNXauw@mail.gmail.com>

Is there any NAT involved? Usually lack of audio means that a NAT router or
firewall is screwing up the RTP media which is sent in a separate UDP
stream. Either it's not being forwarded, or it's being told to go to the
wrong place.

-Steve


On 17 June 2011 13:13, ankIT WALiA <ankitwalia4u at gmail.com> wrote:

> Dear all,
>
> I have configured an gateway for external profile to receive and make
> calls.
>
> I have added a file in public directory.
> <include>
> <extension name="iptel-inbound">
> <condition field="destination_number"
> expression="^(MY_IPTEL_USERNAME)$">
> <action application="set" data="domain_name=$${domain}"/>
> <action application="transfer" data="ExtNo XML default"/>
> </condition>
> </extension>
> </include>
>
> Now, while I am calling to my voip account, I am transferring the call. It
> is getting transferred to the extension also.
> But, If I use normal ext number, I am able to talk both ways. But when I
> use default ext number 5000 for IVR menu. I could not listen to IVR menu.
> Though, I can see in my logs that the audio file are being played one by
> one, but I cant listen.
>
> Do I need some extra configuration to listen to IVR?
> There may be a gap in my knowledge. Please enlighten.
>
> Thanks
> Ankit
> --
> Life is like a rose its upto u feel it as its fragrance or thorns
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Stefan.Weigel at allianz-warranty.com  Fri Jun 17 17:40:52 2011
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Fri, 17 Jun 2011 15:40:52 +0200
Subject: [Freeswitch-users] mod_spandsp event spandsp::rxfaxresult
 variables missing
In-Reply-To: <BANLkTi=-+Ob_F065O-mY8iS819QPJJxw+w@mail.gmail.com>
References: <5003D7D3E06F514E8C682F18D223265C04D3B36C2B@AZWSMS03.azwarranty.int>
	<BANLkTin3vQii2arQd4KhmSBDoJFivaLiBQ@mail.gmail.com>
	<5003D7D3E06F514E8C682F18D223265C04D3B36C2F@AZWSMS03.azwarranty.int>
	<BANLkTi=-+Ob_F065O-mY8iS819QPJJxw+w@mail.gmail.com>
Message-ID: <5003D7D3E06F514E8C682F18D223265C04D3B36C39@AZWSMS03.azwarranty.int>

Hi Steve, list,



I did a fresh 'git clone git://git.freeswitch.org/freeswitch.git<http://git.freeswitch.org/freeswitch.git>' and now I can find the line 'switch_channel_event_set_data(channel, event);' in mod_spandsp_fax.c





Thanks and best regards



Stefan



Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Steven Ayre
Gesendet: Donnerstag, 16. Juni 2011 13:02
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] mod_spandsp event spandsp::rxfaxresult variables missing



Ok. If they're still missing after you rebuild you could reopen the Jira.

-Steve



On 16 June 2011 11:25, Weigel, Stefan <Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>> wrote:

Hi Steve,



version shows me only 'git-.

I'm pretty sure that it was the snapshot downloaded on the 2nd of May. But I can remember also applied the patches.

I checked and the line



                               switch_channel_event_set_data(channel, event);



is missing. I will rebuild the module and test again and tell you more.





Thanks and best regards



Stefan





Von: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] Im Auftrag von Steven Ayre
Gesendet: Donnerstag, 16. Juni 2011 11:47
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] mod_spandsp event spandsp::rxfaxresult variables missing



Do you know the version number of the older working release?



http://jira.freeswitch.org/browse/FS-3004

Looks like Flavio's patch wasn't applied entirely on May 3rd - it's missing the "switch_channel_event_set_data(channel, event);" line. Not sure if that's what's causing it, but it seems likely. The event generation hasn't been modified since then.



-Steve









On 16 June 2011 09:30, Weigel, Stefan <Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>> wrote:

Hi list,



yesterday I had a ODBC related problem. Therefore I used the latest snapshot. Now my scripts are not working properly (I'm doing an ESL connection and listening for spandsp::rxfaxresults events).

With an older release (mid of may) I was getting lot's more info's (for example variable_current_application_data), now I'm only getting the information documented on the wiki page:



fax-document-transferred-pages

fax-document-total-pages

fax-image-resolution

fax-image-size

fax-bad-rows

fax-transfer-rate

fax-result-code

fax-result-text

fax-ecm-used

fax-local-station-id

fax-remote-station-id





What changed and how can I get back the needed information ?







Thanks and best regards



Stefan


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




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



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From linux4michelle at tamay-dogan.net  Fri Jun 17 18:57:14 2011
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Fri, 17 Jun 2011 16:57:14 +0200
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <BANLkTimY5oG_1-Tpk5eBKKC1xX8fXk3AoQ@mail.gmail.com>
References: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
	<201106162131.04859.justlikeef@gmail.com>
	<BANLkTi=5LJDDxzY2d3nirUnQGpOqoBC45Q@mail.gmail.com>
	<BANLkTimY5oG_1-Tpk5eBKKC1xX8fXk3AoQ@mail.gmail.com>
Message-ID: <20110617145714.GX4017@michelle1>

Hello Nandy Dagondon,

Am 2011-06-17 14:07:54, hacktest Du folgendes herunter:
> hello guys!
> 
> i'm trying to setup a small exchange w/ Internet service - of course, using
> FS as the exchange. DSLAMs connect to POTS via 2-wire splitters and that
> would require using FXS gateway (as what Rob mentioned).
> 
> i'm trying to find out if there's a way we can omit the FXS - so it's
> FS-to-DSLAM direct via IP.  i'm not sure if this setup is now possible on
> newer IP DSLAMs.

WHY not give your customers a ADSL/VoIP Gateway where they  connect  the
telephone tothe router?  And then they can connect directly to Asterisk.

Thanks, Greetings and nice Day/Evening
    Michelle Konzack

-- 
##################### Debian GNU/Linux Consultant ######################
   Development of Intranet and Embedded Systems with Debian GNU/Linux

itsystems at tdnet France            itsystems at tdnet
Owner Michelle Konzack            Owner Michelle Konzack

Apt. 917 (homeoffice)             Gewerbe Stra?e 3
50, rue de Soultz                 77694 Kehl/Germany
67100 Strasbourg/France           Tel: +49-177-9351947  mobil
Tel: +33-6-61925193 mobil         Tel: +49-176-86004575 office

<http://www.itsystems.tamay-dogan.net/>  <http://www.flexray4linux.org/>
<http://www.debian.tamay-dogan.net/>         <http://www.can4linux.org/>

Jabber linux4michelle at jabber.ccc.de
ICQ    #328449886

Linux-User #280138 with the Linux Counter, http://counter.li.org/
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From ankitwalia4u at gmail.com  Fri Jun 17 20:07:53 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Fri, 17 Jun 2011 21:37:53 +0530
Subject: [Freeswitch-users] No Voice for default IVR ext 5000
In-Reply-To: <BANLkTinr17B8y2EcwQ0YAVwAGU3giNXauw@mail.gmail.com>
References: <BANLkTi=xZ09f_9HNH6D=c8fpTM72qXbxJA@mail.gmail.com>
	<BANLkTinr17B8y2EcwQ0YAVwAGU3giNXauw@mail.gmail.com>
Message-ID: <BANLkTim7TeR2WxQOCV+-h=CYF-bQPPMUkg@mail.gmail.com>

Yes, I am under NAT through my Broadband router.

But, if NAT is the issue, then is it possible for both sides able to listen
each other voices when I am transferring my call to another ext. which is
not IVR

Could someone please suggest what I can do to work with NAT or under
firewall?

Secondly, do you know any other reason for this issue if its not NAT?

Thanks
Ankit

On Fri, Jun 17, 2011 at 7:05 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Is there any NAT involved? Usually lack of audio means that a NAT router or
> firewall is screwing up the RTP media which is sent in a separate UDP
> stream. Either it's not being forwarded, or it's being told to go to the
> wrong place.
>
> -Steve
>
>
> On 17 June 2011 13:13, ankIT WALiA <ankitwalia4u at gmail.com> wrote:
>
>> Dear all,
>>
>> I have configured an gateway for external profile to receive and make
>> calls.
>>
>> I have added a file in public directory.
>> <include>
>> <extension name="iptel-inbound">
>> <condition field="destination_number"
>> expression="^(MY_IPTEL_USERNAME)$">
>> <action application="set" data="domain_name=$${domain}"/>
>> <action application="transfer" data="ExtNo XML default"/>
>> </condition>
>> </extension>
>> </include>
>>
>> Now, while I am calling to my voip account, I am transferring the call. It
>> is getting transferred to the extension also.
>> But, If I use normal ext number, I am able to talk both ways. But when I
>> use default ext number 5000 for IVR menu. I could not listen to IVR menu.
>> Though, I can see in my logs that the audio file are being played one by
>> one, but I cant listen.
>>
>> Do I need some extra configuration to listen to IVR?
>> There may be a gap in my knowledge. Please enlighten.
>>
>> Thanks
>> Ankit
>> --
>> Life is like a rose its upto u feel it as its fragrance or thorns
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Life is like a rose its upto u feel it as its fragrance or thorns
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From anthony.minessale at gmail.com  Fri Jun 17 21:30:42 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 17 Jun 2011 12:30:42 -0500
Subject: [Freeswitch-users] Want to play music on hold while executing a
 time consuming operation in Lua
In-Reply-To: <BANLkTin2gS3DZWV7Y_DHYB514kEq1drHJA@mail.gmail.com>
References: <BANLkTin5-d2QSCGCaaOW6j+Y_xu3AfNfog@mail.gmail.com>
	<BANLkTikVkBbOJLHQSzuQ4FeqphJnFTaSfw@mail.gmail.com>
	<BANLkTin2gS3DZWV7Y_DHYB514kEq1drHJA@mail.gmail.com>
Message-ID: <BANLkTinw5xw7cgqXF4cWg00_6oQP9Eu93w@mail.gmail.com>

In this case, it turns out you are asking the girl to defy physics for you.

The session is executing the script code in the same thread that needs
to play the audio.
You cannot magically do 2 things at once in the same thread so....
using ASYNC (esl) is your only choice apart from using C and threads
or launching the lua script in another thread with luarun with the
channel uuid as an arg so it can feed it to the constructor and send
commands to it remotely which is a pretty big mess.....




On Thu, Jun 16, 2011 at 4:56 PM, Sidharth Kshatriya
<sid.kshatriya at gmail.com> wrote:
> Thanks Michael. This is quite a mainstream requirement :-) ! Not everyone
> uses ESL. Is there no way we can "hack" this in Lua?
>
> Will await your research,
>
> Sidharth
>
> On Thu, Jun 16, 2011 at 11:51 PM, Michael Collins <msc at freeswitch.org>
> wrote:
>>
>> Actually, I am looking into ways to accomplish these. They are somewhat
>> related to some recipes I am doing for the FS Cookbook. Please stand by.
>> FYI, I am looking at an ESL-based solution, not a dialplan script. Thus far
>> I have not found a viable way to do async stuff like this from a dp script
>> and I'm thinking ESL is the way to go. I will report back in a bit.
>> -MC
>>
>> On Wed, Jun 15, 2011 at 9:53 PM, Sidharth Kshatriya
>> <sid.kshatriya at gmail.com> wrote:
>>>
>>> I am implementing an IVR using Lua in Freeswitch. In my Lua script I use
>>> curl to a web service. Sometimes the response takes a long time to come
>>> back. During that time period I would like to play music on hold.
>>>
>>> I have searched the freeswitch discussion archives a lot. There seem to
>>> be many suggested ways to implement music on hold from a Lua script but the
>>> answers are not very clear / not really applicable to my use case. I don't
>>> know what method I should use and which one is recommended.
>>>
>>> Method 1: Transfer the call to 9664 (music on hold extension). However
>>> the implementation for this solution for this does not seem to be available
>>> in Lua. For example: How would I transfer the call back?
>>> Method 2: Using bgapi uuid_park park the call and using uuid_broadcast
>>> play an audio file. Again what do I do to unpark the call..?
>>>
>>> Thanks,
>>>
>>> Sidharth
>>> [P.S. This is a repost, I apologize but I never got any responses on my
>>> previous email. Need help! :-) ]
>>>
>>> --
>>> Sidharth Kshatriya
>>> www.sidk.info
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sidharth Kshatriya
> www.sidk.info
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From shouldbeq931 at gmail.com  Fri Jun 17 21:33:59 2011
From: shouldbeq931 at gmail.com (shouldbe q931)
Date: Fri, 17 Jun 2011 18:33:59 +0100
Subject: [Freeswitch-users] UK ISDN outgoing CLI (was Re: FreeTDM -
 Sangoma B700 - ISDN connection questions - UK)
In-Reply-To: <BANLkTin5+puYLD7OtreiRxQZ5i8xd5KppQ@mail.gmail.com>
References: <4DED4854.6010203@earthspike.net>
	<BANLkTimP23FVs8Vcg+pbxnfBucrQoJur2w@mail.gmail.com>
	<4DF10C83.6020803@earthspike.net>
	<BANLkTikCEKqPTFZLDxF+h940RnmizHYLBA@mail.gmail.com>
	<4DF1F07A.5000200@earthspike.net>
	<BANLkTin5+puYLD7OtreiRxQZ5i8xd5KppQ@mail.gmail.com>
Message-ID: <BANLkTi=8p2hy6TuPbtQd4tr9oREemVzUhg@mail.gmail.com>

On Mon, Jun 13, 2011 at 7:55 AM, Mike Raistrick
<michael.raistrick at googlemail.com> wrote:
> I've no direct experience with BT BRA ISDN (plenty with other providers) -
> but I don't think that CLIP / COLP will be the issue.
> CLIP is an inbound service - and determines whether your equipment is
> presented with the number of the person calling you (and if they allow it).
> Although COLP is an outbound service, it is used to display the number of
> the line that you are connected through to (which might not be the same as
> the number you dialled - e.g. 0800 numbers).
> Whilst we're on supplementary services, CLIR is also an outbound service and
> allows you to specify whether the CLI that you send (or the one that BT
> uses) can be shown to the person you call.
> Some terminogy: ISDN Basic Rate differentiates Direct Dial In (DDI's) and
> Multiple Subscriber Numbers (MSNs) for incoming services.
> DDIs are designed for PBXs and are associated with a range of numbers in a
> block. The service provider sends you a called party number that can range
> either from the last 'X' digits through to the whole number - this is
> normally specified when the service is ordered and then all calls are
> delivered the same way.
> With DDIs - when you send an outgoing call you should be able to send the
> extension or the whole number - in either case you should mark the Type Of
> Number field accordingly (and if in national format - not include the '0'
> for the number itself). In case that you send the extension digits - the
> service provider adds this to the DDI 'stem' to get the full number. With
> the full number (either derived from the extension, or as you sent it) - the
> provider should then check that the number is in the allocated DDI range -
> if it is it will allow it through, if not it will usually send the root DDI
> number.
> MSNs are unique to BRA - they are completely separate telephone numbers and
> were originally designed to allow multiple BRA phones to be off the same
> ISDN bus and have their own phone numbers. This means that the full number
> will always be sent to you, and be expected from you for called and calling
> numbers (still with the right Type Of Number set..).
> Summary - with either DDI or MSNs - you should be OK to send the whole
> number to BT as the calling party - minus the '0' and with TON set to
> 'national'.
> Finally - if you do have the CLIR supplementary service, make sure that you
> set the presentation indicator to 'allowed' - default should normally be to
> assume allowed unless set differently - but to be on the safe side.
> Mike
>
>

>From my experience with BT, unless you have COLP, you will only send
the lead number of the trunk, you only need CLIR if you want to be
able to restrict


From infos at madovsky.org  Fri Jun 17 21:36:40 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 17 Jun 2011 13:36:40 -0400
Subject: [Freeswitch-users] Want to play music on hold while executing a
	time consuming operation in Lua
References: <BANLkTin5-d2QSCGCaaOW6j+Y_xu3AfNfog@mail.gmail.com><BANLkTikVkBbOJLHQSzuQ4FeqphJnFTaSfw@mail.gmail.com><BANLkTin2gS3DZWV7Y_DHYB514kEq1drHJA@mail.gmail.com>
	<BANLkTinw5xw7cgqXF4cWg00_6oQP9Eu93w@mail.gmail.com>
Message-ID: <4040DFAD059445368E79FE23CC08474C@e1705>

nice metaphore Anth ;)

----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Friday, June 17, 2011 1:30 PM
Subject: Re: [Freeswitch-users] Want to play music on hold while executing a 
time consuming operation in Lua


> In this case, it turns out you are asking the girl to defy physics for 
> you.
>
> The session is executing the script code in the same thread that needs
> to play the audio.
> You cannot magically do 2 things at once in the same thread so....
> using ASYNC (esl) is your only choice apart from using C and threads
> or launching the lua script in another thread with luarun with the
> channel uuid as an arg so it can feed it to the constructor and send
> commands to it remotely which is a pretty big mess.....
>
>
>
>
> On Thu, Jun 16, 2011 at 4:56 PM, Sidharth Kshatriya
> <sid.kshatriya at gmail.com> wrote:
>> Thanks Michael. This is quite a mainstream requirement :-) ! Not everyone
>> uses ESL. Is there no way we can "hack" this in Lua?
>>
>> Will await your research,
>>
>> Sidharth
>>
>> On Thu, Jun 16, 2011 at 11:51 PM, Michael Collins <msc at freeswitch.org>
>> wrote:
>>>
>>> Actually, I am looking into ways to accomplish these. They are somewhat
>>> related to some recipes I am doing for the FS Cookbook. Please stand by.
>>> FYI, I am looking at an ESL-based solution, not a dialplan script. Thus 
>>> far
>>> I have not found a viable way to do async stuff like this from a dp 
>>> script
>>> and I'm thinking ESL is the way to go. I will report back in a bit.
>>> -MC
>>>
>>> On Wed, Jun 15, 2011 at 9:53 PM, Sidharth Kshatriya
>>> <sid.kshatriya at gmail.com> wrote:
>>>>
>>>> I am implementing an IVR using Lua in Freeswitch. In my Lua script I 
>>>> use
>>>> curl to a web service. Sometimes the response takes a long time to come
>>>> back. During that time period I would like to play music on hold.
>>>>
>>>> I have searched the freeswitch discussion archives a lot. There seem to
>>>> be many suggested ways to implement music on hold from a Lua script but 
>>>> the
>>>> answers are not very clear / not really applicable to my use case. I 
>>>> don't
>>>> know what method I should use and which one is recommended.
>>>>
>>>> Method 1: Transfer the call to 9664 (music on hold extension). However
>>>> the implementation for this solution for this does not seem to be 
>>>> available
>>>> in Lua. For example: How would I transfer the call back?
>>>> Method 2: Using bgapi uuid_park park the call and using uuid_broadcast
>>>> play an audio file. Again what do I do to unpark the call..?
>>>>
>>>> Thanks,
>>>>
>>>> Sidharth
>>>> [P.S. This is a repost, I apologize but I never got any responses on my
>>>> previous email. Need help! :-) ]
>>>>
>>>> --
>>>> Sidharth Kshatriya
>>>> www.sidk.info
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Sidharth Kshatriya
>> www.sidk.info
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From msc at freeswitch.org  Fri Jun 17 21:44:04 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 17 Jun 2011 10:44:04 -0700
Subject: [Freeswitch-users] Want to play music on hold while executing a
 time consuming operation in Lua
In-Reply-To: <BANLkTinw5xw7cgqXF4cWg00_6oQP9Eu93w@mail.gmail.com>
References: <BANLkTin5-d2QSCGCaaOW6j+Y_xu3AfNfog@mail.gmail.com>
	<BANLkTikVkBbOJLHQSzuQ4FeqphJnFTaSfw@mail.gmail.com>
	<BANLkTin2gS3DZWV7Y_DHYB514kEq1drHJA@mail.gmail.com>
	<BANLkTinw5xw7cgqXF4cWg00_6oQP9Eu93w@mail.gmail.com>
Message-ID: <BANLkTimWe-vEHjHZRvoTT+ZcEFPOVttVuQ@mail.gmail.com>

Tony,

Thanks for the explanation. This was indeed my concern with trying to do it
from a dp script. Sid, I'm afraid you'll need to use ESL or find a girl who
is not bound by the laws of physics... :)

-MC

On Fri, Jun 17, 2011 at 10:30 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> In this case, it turns out you are asking the girl to defy physics for you.
>
> The session is executing the script code in the same thread that needs
> to play the audio.
> You cannot magically do 2 things at once in the same thread so....
> using ASYNC (esl) is your only choice apart from using C and threads
> or launching the lua script in another thread with luarun with the
> channel uuid as an arg so it can feed it to the constructor and send
> commands to it remotely which is a pretty big mess.....
>
>
>
>
> On Thu, Jun 16, 2011 at 4:56 PM, Sidharth Kshatriya
> <sid.kshatriya at gmail.com> wrote:
> > Thanks Michael. This is quite a mainstream requirement :-) ! Not everyone
> > uses ESL. Is there no way we can "hack" this in Lua?
> >
> > Will await your research,
> >
> > Sidharth
> >
> > On Thu, Jun 16, 2011 at 11:51 PM, Michael Collins <msc at freeswitch.org>
> > wrote:
> >>
> >> Actually, I am looking into ways to accomplish these. They are somewhat
> >> related to some recipes I am doing for the FS Cookbook. Please stand by.
> >> FYI, I am looking at an ESL-based solution, not a dialplan script. Thus
> far
> >> I have not found a viable way to do async stuff like this from a dp
> script
> >> and I'm thinking ESL is the way to go. I will report back in a bit.
> >> -MC
> >>
> >> On Wed, Jun 15, 2011 at 9:53 PM, Sidharth Kshatriya
> >> <sid.kshatriya at gmail.com> wrote:
> >>>
> >>> I am implementing an IVR using Lua in Freeswitch. In my Lua script I
> use
> >>> curl to a web service. Sometimes the response takes a long time to come
> >>> back. During that time period I would like to play music on hold.
> >>>
> >>> I have searched the freeswitch discussion archives a lot. There seem to
> >>> be many suggested ways to implement music on hold from a Lua script but
> the
> >>> answers are not very clear / not really applicable to my use case. I
> don't
> >>> know what method I should use and which one is recommended.
> >>>
> >>> Method 1: Transfer the call to 9664 (music on hold extension). However
> >>> the implementation for this solution for this does not seem to be
> available
> >>> in Lua. For example: How would I transfer the call back?
> >>> Method 2: Using bgapi uuid_park park the call and using uuid_broadcast
> >>> play an audio file. Again what do I do to unpark the call..?
> >>>
> >>> Thanks,
> >>>
> >>> Sidharth
> >>> [P.S. This is a repost, I apologize but I never got any responses on my
> >>> previous email. Need help! :-) ]
> >>>
> >>> --
> >>> Sidharth Kshatriya
> >>> www.sidk.info
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > --
> > Sidharth Kshatriya
> > www.sidk.info
> >
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From curriegrad2004 at gmail.com  Fri Jun 17 21:46:06 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Fri, 17 Jun 2011 10:46:06 -0700
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <20110617145714.GX4017@michelle1>
References: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
	<201106162131.04859.justlikeef@gmail.com>
	<BANLkTi=5LJDDxzY2d3nirUnQGpOqoBC45Q@mail.gmail.com>
	<BANLkTimY5oG_1-Tpk5eBKKC1xX8fXk3AoQ@mail.gmail.com>
	<20110617145714.GX4017@michelle1>
Message-ID: <BANLkTimoajshNrVJgdg5kmGbhDBLjW31aQ@mail.gmail.com>

Actually, most of his customers are still in the dark ages... That's
the problem here ;)

On Fri, Jun 17, 2011 at 7:57 AM, Michelle Konzack
<linux4michelle at tamay-dogan.net> wrote:
> Hello Nandy Dagondon,
>
> Am 2011-06-17 14:07:54, hacktest Du folgendes herunter:
>> hello guys!
>>
>> i'm trying to setup a small exchange w/ Internet service - of course, using
>> FS as the exchange. DSLAMs connect to POTS via 2-wire splitters and that
>> would require using FXS gateway (as what Rob mentioned).
>>
>> i'm trying to find out if there's a way we can omit the FXS - so it's
>> FS-to-DSLAM direct via IP. ?i'm not sure if this setup is now possible on
>> newer IP DSLAMs.
>
> WHY not give your customers a ADSL/VoIP Gateway where they ?connect ?the
> telephone tothe router? ?And then they can connect directly to Asterisk.
>
> Thanks, Greetings and nice Day/Evening
> ? ?Michelle Konzack
>
> --
> ##################### Debian GNU/Linux Consultant ######################
> ? Development of Intranet and Embedded Systems with Debian GNU/Linux
>
> itsystems at tdnet France ? ? ? ? ? ?itsystems at tdnet
> Owner Michelle Konzack ? ? ? ? ? ?Owner Michelle Konzack
>
> Apt. 917 (homeoffice) ? ? ? ? ? ? Gewerbe Stra?e 3
> 50, rue de Soultz ? ? ? ? ? ? ? ? 77694 Kehl/Germany
> 67100 Strasbourg/France ? ? ? ? ? Tel: +49-177-9351947 ?mobil
> Tel: +33-6-61925193 mobil ? ? ? ? Tel: +49-176-86004575 office
>
> <http://www.itsystems.tamay-dogan.net/> ?<http://www.flexray4linux.org/>
> <http://www.debian.tamay-dogan.net/> ? ? ? ? <http://www.can4linux.org/>
>
> Jabber linux4michelle at jabber.ccc.de
> ICQ ? ?#328449886
>
> Linux-User #280138 with the Linux Counter, http://counter.li.org/
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From freeswitch at earthspike.net  Fri Jun 17 22:08:49 2011
From: freeswitch at earthspike.net (John)
Date: Fri, 17 Jun 2011 19:08:49 +0100
Subject: [Freeswitch-users] UK ISDN outgoing CLI (was Re: FreeTDM -
 Sangoma B700 - ISDN connection questions - UK)
In-Reply-To: <BANLkTi=8p2hy6TuPbtQd4tr9oREemVzUhg@mail.gmail.com>
References: <4DED4854.6010203@earthspike.net>	<BANLkTimP23FVs8Vcg+pbxnfBucrQoJur2w@mail.gmail.com>	<4DF10C83.6020803@earthspike.net>	<BANLkTikCEKqPTFZLDxF+h940RnmizHYLBA@mail.gmail.com>	<4DF1F07A.5000200@earthspike.net>	<BANLkTin5+puYLD7OtreiRxQZ5i8xd5KppQ@mail.gmail.com>
	<BANLkTi=8p2hy6TuPbtQd4tr9oREemVzUhg@mail.gmail.com>
Message-ID: <4DFB9831.4040709@earthspike.net>

Mike, Shouldbe,

We now have CLIP on the line for sure, and I ordered COLP at the same 
time and am assured that that has been turned on as well, but I still 
cannot get outgoing CLI to work properly.  Mike, you mention setting the 
TON to 'national'; where is that setting?

At the moment, my dialplan looks like 
http://pastebin.freeswitch.org/16548 with .  An outgoing call gives the 
log 7 at http://pastebin.freeswitch.org/16549 (numbers have been changed 
to protect the innocent!) but the number that 01234 567890 sees on their 
Caller ID is not 876543 but the main number of the line (876540, say).  
I have tried setting the outgoing_caller_id_number to 6, 10 and 11 
digits, restarting FS after each change, but with no success.

The service provider has only one clue to offer: "With regards to the 
configuration at the exchange, the line is set to 6 digits to switch." 
which makes eminent sense, and suggests that I should be presenting 6 
digits.  Incoming calls have a 6 digit called number and 10 digit 
calling number.

Thanks for the help so far.  Any other ideas?

John


On 17/06/11 18:33, shouldbe q931 wrote:

> From my experience with BT, unless you have COLP, you will only send
the lead number of the trunk, you only need CLIR if you want to be
able to restrict


> On Mon, Jun 13, 2011 at 7:55 AM, Mike Raistrick
> <michael.raistrick at googlemail.com>  wrote:
>> I've no direct experience with BT BRA ISDN (plenty with other providers) -
>> but I don't think that CLIP / COLP will be the issue.
>> CLIP is an inbound service - and determines whether your equipment is
>> presented with the number of the person calling you (and if they allow it).
>> Although COLP is an outbound service, it is used to display the number of
>> the line that you are connected through to (which might not be the same as
>> the number you dialled - e.g. 0800 numbers).
>> Whilst we're on supplementary services, CLIR is also an outbound service and
>> allows you to specify whether the CLI that you send (or the one that BT
>> uses) can be shown to the person you call.
>> Some terminogy: ISDN Basic Rate differentiates Direct Dial In (DDI's) and
>> Multiple Subscriber Numbers (MSNs) for incoming services.
>> DDIs are designed for PBXs and are associated with a range of numbers in a
>> block. The service provider sends you a called party number that can range
>> either from the last 'X' digits through to the whole number - this is
>> normally specified when the service is ordered and then all calls are
>> delivered the same way.
>> With DDIs - when you send an outgoing call you should be able to send the
>> extension or the whole number - in either case you should mark the Type Of
>> Number field accordingly (and if in national format - not include the '0'
>> for the number itself). In case that you send the extension digits - the
>> service provider adds this to the DDI 'stem' to get the full number. With
>> the full number (either derived from the extension, or as you sent it) - the
>> provider should then check that the number is in the allocated DDI range -
>> if it is it will allow it through, if not it will usually send the root DDI
>> number.
>> MSNs are unique to BRA - they are completely separate telephone numbers and
>> were originally designed to allow multiple BRA phones to be off the same
>> ISDN bus and have their own phone numbers. This means that the full number
>> will always be sent to you, and be expected from you for called and calling
>> numbers (still with the right Type Of Number set..).
>> Summary - with either DDI or MSNs - you should be OK to send the whole
>> number to BT as the calling party - minus the '0' and with TON set to
>> 'national'.
>> Finally - if you do have the CLIR supplementary service, make sure that you
>> set the presentation indicator to 'allowed' - default should normally be to
>> assume allowed unless set differently - but to be on the safe side.
>> Mike
>>
>>

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From freeswitch at peely.com  Fri Jun 17 22:12:23 2011
From: freeswitch at peely.com (peely)
Date: Fri, 17 Jun 2011 11:12:23 -0700 (PDT)
Subject: [Freeswitch-users] Can't make mod_rtmp
Message-ID: <1308334343913-6488229.post@n2.nabble.com>

Hi,

I'm having trouble building FreeSWITCH with the latest mod_rtmp goodness. On
Ubuntu X64 compiling with gcc I get:

cc1: warnings being treated as errors
/usr/src/freeswitch/src/mod/endpoints/mod_rtmp/mod_rtmp.c: In function
?rtmp_function?:
/usr/src/freeswitch/src/mod/endpoints/mod_rtmp/mod_rtmp.c:1612:114: error:
comparison between ?switch_call_cause_t? and ?enum <anonymous>?


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


Thanks,



Neil.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229p6488229.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From shouldbeq931 at gmail.com  Fri Jun 17 22:12:34 2011
From: shouldbeq931 at gmail.com (shouldbe q931)
Date: Fri, 17 Jun 2011 19:12:34 +0100
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <BANLkTinLho7z+XgBHGjUJY7+vyaWV60NMA@mail.gmail.com>
References: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
	<201106162131.04859.justlikeef@gmail.com>
	<BANLkTi=5LJDDxzY2d3nirUnQGpOqoBC45Q@mail.gmail.com>
	<BANLkTimY5oG_1-Tpk5eBKKC1xX8fXk3AoQ@mail.gmail.com>
	<C201E991-C3A1-437E-BC22-C82FD7360FC1@ipeva.fr>
	<BANLkTin5YSNPPkQfLNuht=yTxmgKeRWAVw@mail.gmail.com>
	<45FF3438-C239-48D9-A57B-F0A49166900E@ipeva.fr>
	<BANLkTin3apQVJLXh-novL0deaq+QM0TuTQ@mail.gmail.com>
	<0013F8A2-8D82-470C-B8D8-45C22BDB3A00@ipeva.fr>
	<BANLkTinLho7z+XgBHGjUJY7+vyaWV60NMA@mail.gmail.com>
Message-ID: <BANLkTimSqd+XqBQMh5HtmxCfKx+1hZKvvg@mail.gmail.com>

On Fri, Jun 17, 2011 at 11:57 AM, Nandy Dagondon <gcd at i.ph> wrote:
> huh ... that's quiet pricey. perhaps an IP DSLAM + FXS would be a cheaper
> setup. in rural areas, many are still dinosaurs.
>
>
Not a clue about price, but Zyxel and Draytek make some "small"
DSLAM/MSAN units, Draytek also make some DSL routers with integrated
FXS ports, note sure if they would interop with Freswitch...


From shouldbeq931 at gmail.com  Fri Jun 17 22:43:35 2011
From: shouldbeq931 at gmail.com (shouldbe q931)
Date: Fri, 17 Jun 2011 19:43:35 +0100
Subject: [Freeswitch-users] UK ISDN outgoing CLI (was Re: FreeTDM -
 Sangoma B700 - ISDN connection questions - UK)
In-Reply-To: <4DFB9831.4040709@earthspike.net>
References: <4DED4854.6010203@earthspike.net>
	<BANLkTimP23FVs8Vcg+pbxnfBucrQoJur2w@mail.gmail.com>
	<4DF10C83.6020803@earthspike.net>
	<BANLkTikCEKqPTFZLDxF+h940RnmizHYLBA@mail.gmail.com>
	<4DF1F07A.5000200@earthspike.net>
	<BANLkTin5+puYLD7OtreiRxQZ5i8xd5KppQ@mail.gmail.com>
	<BANLkTi=8p2hy6TuPbtQd4tr9oREemVzUhg@mail.gmail.com>
	<4DFB9831.4040709@earthspike.net>
Message-ID: <BANLkTinXZo=e-Cd+uL=mX21jXZNYBWdHuA@mail.gmail.com>

On Fri, Jun 17, 2011 at 7:08 PM, John <freeswitch at earthspike.net> wrote:
> Mike, Shouldbe,
>
> We now have CLIP on the line for sure, and I ordered COLP at the same time
> and am assured that that has been turned on as well, but I still cannot get
> outgoing CLI to work properly.? Mike, you mention setting the TON to
> 'national'; where is that setting?
>
> At the moment, my dialplan looks like http://pastebin.freeswitch.org/16548
> with .? An outgoing call gives the log 7 at
> http://pastebin.freeswitch.org/16549 (numbers have been changed to protect
> the innocent!) but the number that 01234 567890 sees on their Caller ID is
> not 876543 but the main number of the line (876540, say).? I have tried
> setting the outgoing_caller_id_number to 6, 10 and 11 digits, restarting FS
> after each change, but with no success.
>
> The service provider has only one clue to offer: "With regards to the
> configuration at the exchange, the line is set to 6 digits to switch." which
> makes eminent sense, and suggests that I should be presenting 6 digits.
> Incoming calls have a 6 digit called number and 10 digit calling number.
>
> Thanks for the help so far.? Any other ideas?
>
> John
>
>

I don't have any experience with FS connecting over ISDN so can't help
you much further on the FS configuration:-(

However on the lines, I would be very tempted to see if you can borrow
a BRI tester, or what I've frequently used in the past is an
Eicon/Dialogic card as the test functions (under windows) are nearly
as good, that might help you see if it _is_ a BT problem, or a FS
configuration issue. It would need a 2k/XP machine, but something like
http://cgi.ebay.co.uk/EICON-DIVA-2-01-PCI-GRAPHICS-CARD-/270761571564
is what I have used in the past. I think I might have one of the
PCMCIA ones somewhere, but I'd need to test it still works

I'm fairly sure that the "6 digits to switch" is what they are sending
to you, not what they are expecting from you. I can't access out
remaining BRI (Avaya) PBX from here, but on our PRI (again Avaya)
switches, we are sending 10 digits.

Cheers


From msc at freeswitch.org  Fri Jun 17 23:05:55 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 17 Jun 2011 12:05:55 -0700
Subject: [Freeswitch-users] No Voice for default IVR ext 5000
In-Reply-To: <BANLkTim7TeR2WxQOCV+-h=CYF-bQPPMUkg@mail.gmail.com>
References: <BANLkTi=xZ09f_9HNH6D=c8fpTM72qXbxJA@mail.gmail.com>
	<BANLkTinr17B8y2EcwQ0YAVwAGU3giNXauw@mail.gmail.com>
	<BANLkTim7TeR2WxQOCV+-h=CYF-bQPPMUkg@mail.gmail.com>
Message-ID: <BANLkTikrkZ80n6R_JwyGq1T3RvbG5RrWQQ@mail.gmail.com>

I would get a pcap of both the signaling and the RTP and analyze w/
wireshark. Find out if the RTP is actually leaving FreeSWITCH or not. Of
course, you might start with a console debug log and siptrace on
pastebin.freeswitch.org so that we can have a looksee at what is happening
on the call.

-MC

On Fri, Jun 17, 2011 at 9:07 AM, ankIT WALiA <ankitwalia4u at gmail.com> wrote:

> Yes, I am under NAT through my Broadband router.
>
> But, if NAT is the issue, then is it possible for both sides able to listen
> each other voices when I am transferring my call to another ext. which is
> not IVR
>
> Could someone please suggest what I can do to work with NAT or under
> firewall?
>
> Secondly, do you know any other reason for this issue if its not NAT?
>
> Thanks
> Ankit
>
>
> On Fri, Jun 17, 2011 at 7:05 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Is there any NAT involved? Usually lack of audio means that a NAT router
>> or firewall is screwing up the RTP media which is sent in a separate UDP
>> stream. Either it's not being forwarded, or it's being told to go to the
>> wrong place.
>>
>> -Steve
>>
>>
>> On 17 June 2011 13:13, ankIT WALiA <ankitwalia4u at gmail.com> wrote:
>>
>>> Dear all,
>>>
>>> I have configured an gateway for external profile to receive and make
>>> calls.
>>>
>>> I have added a file in public directory.
>>> <include>
>>> <extension name="iptel-inbound">
>>> <condition field="destination_number"
>>> expression="^(MY_IPTEL_USERNAME)$">
>>> <action application="set" data="domain_name=$${domain}"/>
>>> <action application="transfer" data="ExtNo XML default"/>
>>> </condition>
>>> </extension>
>>> </include>
>>>
>>> Now, while I am calling to my voip account, I am transferring the call.
>>> It is getting transferred to the extension also.
>>> But, If I use normal ext number, I am able to talk both ways. But when I
>>> use default ext number 5000 for IVR menu. I could not listen to IVR menu.
>>> Though, I can see in my logs that the audio file are being played one by
>>> one, but I cant listen.
>>>
>>> Do I need some extra configuration to listen to IVR?
>>> There may be a gap in my knowledge. Please enlighten.
>>>
>>> Thanks
>>> Ankit
>>> --
>>> Life is like a rose its upto u feel it as its fragrance or thorns
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Life is like a rose its upto u feel it as its fragrance or thorns
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From abdmeziane at gmail.com  Fri Jun 17 22:34:55 2011
From: abdmeziane at gmail.com (abderrahmane abdmeziane)
Date: Fri, 17 Jun 2011 20:34:55 +0200
Subject: [Freeswitch-users] Registration problem with default extension
Message-ID: <BANLkTikC3GjLw4iwO6L__ZgypB5U9ZqGcA@mail.gmail.com>

hey,after installing freeswitch from deb package,and using default extension
1000 with default password 1234 in xlite softphone,i can't call another
extension just don't ring but in xlite screen i get my user name with my
number;a check the output of F9 from fs_cli command,but there is no register
extension:
##############################################
freeswitch at internal> sofia status profile internal

=================================================================================================
Name                    internal
Domain Name             N/A
Auto-NAT                true
DBName                  sofia_reg_internal
Pres Hosts              192.168.2.2,192.168.2.2
Dialplan                XML
Context                 public
Challenge Realm         auto_from
RTP-IP                  192.168.2.2
Ext-RTP-IP              41.104.52.52
SIP-IP                  192.168.2.2
Ext-SIP-IP              41.104.52.52
URL                     sip:mod_sofia at 192.168.2.2:5060
BIND-URL                sip:mod_sofia at 192.168.2.2:5060
HOLD-MUSIC              local_stream://moh
OUTBOUND-PROXY          N/A
CODECS IN               G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
CODECS OUT              G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
TEL-EVENT               101
DTMF-MODE               rfc2833
CNG                     13
SESSION-TO              0
MAX-DIALOG              0
NOMEDIA                 false
LATE-NEG                false
PROXY-MEDIA             false
AGGRESSIVENAT           false
STUN-ENABLED            true
STUN-AUTO-DISABLE       false
CALLS-IN                0
FAILED-CALLS-IN         0
CALLS-OUT               0
FAILED-CALLS-OUT        0

Registrations:
=================================================================================================
Total items returned: 0
=================================================================================================

freeswitch at internal>
#################################
any help please?
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From anthony.minessale at gmail.com  Fri Jun 17 23:20:25 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 17 Jun 2011 14:20:25 -0500
Subject: [Freeswitch-users] Barracuda Networks Donates mod_rtmp for
	FreeSWITCH (Flash Audio Endpoint)
Message-ID: <BANLkTik6DoN=xjbYU8WVyS8Gy1qTihNOoA@mail.gmail.com>

check it out

http://slashdot.org/submission/1657690/Barracuda-Networks-Connects-FLASH-To-SIP-for-FREE
mark it worthy and rate it

http://digg.com/news/entertainment/barracuda_networks_connects_flash_to_sip_for_free_2
digg it



-- 
Anthony Minessale II

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

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

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


From msc at freeswitch.org  Fri Jun 17 23:34:19 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 17 Jun 2011 12:34:19 -0700
Subject: [Freeswitch-users] Generating ringback if 180 arrives after 183
In-Reply-To: <1f3159c1-33ac-42f1-8277-fcb0b79829c7@mx.voismart.com>
References: <1f3159c1-33ac-42f1-8277-fcb0b79829c7@mx.voismart.com>
Message-ID: <BANLkTinSBBC3icB5UMYiA5O4jCetegMxLA@mail.gmail.com>

Any chance you can find the person who designed their SIP stack and hit him
with a ClueBat?
-MC

On Fri, Jun 17, 2011 at 4:14 AM, Matteo <mbrancaleoni at voismart.it> wrote:

> Hi all,
>
> I've the following scenario from my upstream provider :
>
> FS calls provider
> Provider sends easrly media with 183 announcing that the call is going to
> be trasferred
> Provider sends 180 without sdp
>
> In my dialplan I'm using ringback variable to let FS generate ringback
> to internal network, and if I receive 180 without 183 is working ok.
>
> What I'm trying to achieve is to let ringback work even if we've already
> received
> the 183 with sdp.
>
> I've tryed a lot of dialplan magic without luck.
> Maybe we need to add this support into FS itself ?
>
> What I see seems that we need to break the bridge or at least
> "inject" the tone stream into the internal leg of the call.
>
> To be fair, the exact sequence is:
>
> FS calls provider
> provider sends 183/sdp annoucing the call trasnfer
> provider sends update which stops the media (media=inactive)
> provider sends 180 ringing (we must generate here)
>
> maybe the correct way is to handle the update, put fs back
> into ringing state and proceed from there?
> Is feasible or we can get the same result with some dialplan magic?
>
> any hint is appreciated :)
>
> regards,
> matteo
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Sat Jun 18 00:05:45 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 17 Jun 2011 15:05:45 -0500
Subject: [Freeswitch-users] Can't make mod_rtmp
In-Reply-To: <1308334343913-6488229.post@n2.nabble.com>
References: <1308334343913-6488229.post@n2.nabble.com>
Message-ID: <BANLkTimx1MnAhfhHVmVds3qYpqNpaC9JKQ@mail.gmail.com>

update and try again

On Fri, Jun 17, 2011 at 1:12 PM, peely <freeswitch at peely.com> wrote:
> Hi,
>
> I'm having trouble building FreeSWITCH with the latest mod_rtmp goodness. On
> Ubuntu X64 compiling with gcc I get:
>
> cc1: warnings being treated as errors
> /usr/src/freeswitch/src/mod/endpoints/mod_rtmp/mod_rtmp.c: In function
> ?rtmp_function?:
> /usr/src/freeswitch/src/mod/endpoints/mod_rtmp/mod_rtmp.c:1612:114: error:
> comparison between ?switch_call_cause_t? and ?enum <anonymous>?
>
>
> I'm not sure if I need to do anything else, I just naively added
> endpoints/mod_rtmp to modules.conf and did a make clean install?
>
>
> Thanks,
>
>
>
> Neil.
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229p6488229.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From david.ponzone at ipeva.fr  Sat Jun 18 00:16:06 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 17 Jun 2011 22:16:06 +0200
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <BANLkTimSqd+XqBQMh5HtmxCfKx+1hZKvvg@mail.gmail.com>
References: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
	<201106162131.04859.justlikeef@gmail.com>
	<BANLkTi=5LJDDxzY2d3nirUnQGpOqoBC45Q@mail.gmail.com>
	<BANLkTimY5oG_1-Tpk5eBKKC1xX8fXk3AoQ@mail.gmail.com>
	<C201E991-C3A1-437E-BC22-C82FD7360FC1@ipeva.fr>
	<BANLkTin5YSNPPkQfLNuht=yTxmgKeRWAVw@mail.gmail.com>
	<45FF3438-C239-48D9-A57B-F0A49166900E@ipeva.fr>
	<BANLkTin3apQVJLXh-novL0deaq+QM0TuTQ@mail.gmail.com>
	<0013F8A2-8D82-470C-B8D8-45C22BDB3A00@ipeva.fr>
	<BANLkTinLho7z+XgBHGjUJY7+vyaWV60NMA@mail.gmail.com>
	<BANLkTimSqd+XqBQMh5HtmxCfKx+1hZKvvg@mail.gmail.com>
Message-ID: <54DA215E-D28E-4CCB-9B86-A8DAF1E6BCBA@ipeva.fr>

Draytek gear is crap like hell.
Stay away from that.
The support is crap too.

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

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

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




Le 17/06/2011 ? 20:12, shouldbe q931 a ?crit :

> On Fri, Jun 17, 2011 at 11:57 AM, Nandy Dagondon <gcd at i.ph> wrote:
>> huh ... that's quiet pricey. perhaps an IP DSLAM + FXS would be a cheaper
>> setup. in rural areas, many are still dinosaurs.
>> 
>> 
> Not a clue about price, but Zyxel and Draytek make some "small"
> DSLAM/MSAN units, Draytek also make some DSL routers with integrated
> FXS ports, note sure if they would interop with Freswitch...
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From grsingh750 at gmail.com  Sat Jun 18 00:28:44 2011
From: grsingh750 at gmail.com (guru singh)
Date: Sat, 18 Jun 2011 01:58:44 +0530
Subject: [Freeswitch-users] No Voice for default IVR ext 5000
In-Reply-To: <BANLkTikrkZ80n6R_JwyGq1T3RvbG5RrWQQ@mail.gmail.com>
References: <BANLkTi=xZ09f_9HNH6D=c8fpTM72qXbxJA@mail.gmail.com>
	<BANLkTinr17B8y2EcwQ0YAVwAGU3giNXauw@mail.gmail.com>
	<BANLkTim7TeR2WxQOCV+-h=CYF-bQPPMUkg@mail.gmail.com>
	<BANLkTikrkZ80n6R_JwyGq1T3RvbG5RrWQQ@mail.gmail.com>
Message-ID: <BANLkTin8_p2H5MMZK+U7vC4vPnp7=46atg@mail.gmail.com>

Hi Ankit,

You could also see if there are SIP ALG settings on the router.

If by broadband router you mean the DSL modem that providers give,
then you're in a sticky situation. From my experience they are very
unreliable, something that works would break without any apparent
reason or even worse at times things would only work intermittently.
So you just end up racking your brains trying to reproduce and
diagnose, what I'd call "a false problem". Replacing that would reduce
some of your troubles.

Regards,
guru

On Sat, Jun 18, 2011 at 12:35 AM, Michael Collins <msc at freeswitch.org> wrote:
> I would get a pcap of both the signaling and the RTP and analyze w/
> wireshark. Find out if the RTP is actually leaving FreeSWITCH or not. Of
> course, you might start with a console debug log and siptrace on
> pastebin.freeswitch.org so that we can have a looksee at what is happening
> on the call.
> -MC
>
> On Fri, Jun 17, 2011 at 9:07 AM, ankIT WALiA <ankitwalia4u at gmail.com> wrote:
>>
>> Yes, I am under NAT through my Broadband router.
>> But, if NAT is the issue, then is it possible for?both sides able to
>> listen each other voices?when I am?transferring?my call to another ext.
>> which is not IVR
>> Could someone please suggest what I can do to work with NAT or under
>> firewall?
>> Secondly, do you know any other reason for this issue if its not NAT?
>> Thanks
>> Ankit
>>
>> On Fri, Jun 17, 2011 at 7:05 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>>
>>> Is there any NAT involved? Usually lack of audio means that a NAT router
>>> or firewall is screwing up the RTP media which is sent in a separate UDP
>>> stream. Either it's not being forwarded, or it's being told to go to the
>>> wrong place.
>>>
>>> -Steve
>>>
>>>
>>> On 17 June 2011 13:13, ankIT WALiA <ankitwalia4u at gmail.com> wrote:
>>>>
>>>> Dear all,
>>>> I have configured an gateway for external profile to receive and make
>>>> calls.
>>>> I have added a file in public directory.
>>>> <include>
>>>> <extension name="iptel-inbound">
>>>> <condition field="destination_number"
>>>> expression="^(MY_IPTEL_USERNAME)$">
>>>> <action application="set" data="domain_name=$${domain}"/>
>>>> <action application="transfer" data="ExtNo XML default"/>
>>>> </condition>
>>>> </extension>
>>>> </include>
>>>> Now, while I am calling to my voip account, I am?transferring?the call.
>>>> It is getting transferred to the extension also.
>>>> But, If I use normal ext number, I am able to talk both ways. But when I
>>>> use default ext number 5000 for IVR menu. I could not listen to IVR menu.
>>>> Though, I can see in my logs that the audio file are being played one by
>>>> one, but I cant listen.
>>>> Do I need some extra configuration to listen to IVR?
>>>> There may be a gap in my knowledge. Please enlighten.
>>>> Thanks
>>>> Ankit
>>>> --
>>>> Life is like a rose its upto u feel it as its fragrance or thorns
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Life is like a rose its upto u feel it as its fragrance or thorns
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From mbrancaleoni at voismart.it  Sat Jun 18 00:43:23 2011
From: mbrancaleoni at voismart.it (Matteo)
Date: Fri, 17 Jun 2011 22:43:23 +0200 (CEST)
Subject: [Freeswitch-users] Generating ringback if 180 arrives after 183
In-Reply-To: <BANLkTinSBBC3icB5UMYiA5O4jCetegMxLA@mail.gmail.com>
Message-ID: <7d1e7eba-6248-44e5-bfee-3723369e1784@mx.voismart.com>

Hi,

----- Messaggio originale -----
> 
> Any chance you can find the person who designed their SIP stack and
> hit him with a ClueBat?

why? I think that can be correct, even on sip implementors someone expresses
that if we receive 180 after 183 local ringing must be generated.
And this is "more right" if before the 180 an update which tells us
to stop the media is sent, like in this case.

or I'm missing something?

mat


From msc at freeswitch.org  Sat Jun 18 00:42:33 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 17 Jun 2011 13:42:33 -0700
Subject: [Freeswitch-users] Registration problem with default extension
In-Reply-To: <BANLkTikC3GjLw4iwO6L__ZgypB5U9ZqGcA@mail.gmail.com>
References: <BANLkTikC3GjLw4iwO6L__ZgypB5U9ZqGcA@mail.gmail.com>
Message-ID: <BANLkTinaZVgzKv3CorrWiV1784BU31cGjg@mail.gmail.com>

Most likely there is a setting wrong in the x-lite config. If you watch the
freeswitch console you will see any registration attempts. If you want to
see what's REALLY happening then enable SIP trace:

sofia profile internal siptrace on

Then watch to see what traffic passes between x-lite and freeswitch. Look
for messages like "403 Forbidden" which means the password is wrong. If you
see no registration attempts from the phone then that means there's some
network issue or the IP address is not set correctly in the x-lite config.

-MC

On Fri, Jun 17, 2011 at 11:34 AM, abderrahmane abdmeziane <
abdmeziane at gmail.com> wrote:

> hey,after installing freeswitch from deb package,and using default
> extension 1000 with default password 1234 in xlite softphone,i can't call
> another extension just don't ring but in xlite screen i get my user name
> with my number;a check the output of F9 from fs_cli command,but there is no
> register extension:
> ##############################################
> freeswitch at internal> sofia status profile internal
>
>
> =================================================================================================
> Name                    internal
> Domain Name             N/A
> Auto-NAT                true
> DBName                  sofia_reg_internal
> Pres Hosts              192.168.2.2,192.168.2.2
> Dialplan                XML
> Context                 public
> Challenge Realm         auto_from
> RTP-IP                  192.168.2.2
> Ext-RTP-IP              41.104.52.52
> SIP-IP                  192.168.2.2
> Ext-SIP-IP              41.104.52.52
> URL                     sip:mod_sofia at 192.168.2.2:5060
> BIND-URL                sip:mod_sofia at 192.168.2.2:5060
> HOLD-MUSIC              local_stream://moh
> OUTBOUND-PROXY          N/A
> CODECS IN               G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
> CODECS OUT              G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
> TEL-EVENT               101
> DTMF-MODE               rfc2833
> CNG                     13
> SESSION-TO              0
> MAX-DIALOG              0
> NOMEDIA                 false
> LATE-NEG                false
> PROXY-MEDIA             false
> AGGRESSIVENAT           false
> STUN-ENABLED            true
> STUN-AUTO-DISABLE       false
> CALLS-IN                0
> FAILED-CALLS-IN         0
> CALLS-OUT               0
> FAILED-CALLS-OUT        0
>
> Registrations:
>
> =================================================================================================
> Total items returned: 0
>
> =================================================================================================
>
> freeswitch at internal>
> #################################
> any help please?
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gcd at i.ph  Sat Jun 18 01:54:23 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Sat, 18 Jun 2011 05:54:23 +0800
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <20110617145714.GX4017@michelle1>
References: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
	<201106162131.04859.justlikeef@gmail.com>
	<BANLkTi=5LJDDxzY2d3nirUnQGpOqoBC45Q@mail.gmail.com>
	<BANLkTimY5oG_1-Tpk5eBKKC1xX8fXk3AoQ@mail.gmail.com>
	<20110617145714.GX4017@michelle1>
Message-ID: <BANLkTi=GKEKqmSQzKr9KSsXLGfRHUOut4g@mail.gmail.com>

actually this is what we need if the subscriber needs Internet connection.
any suggested brand/model of ADSL/VoIP gateway? that would be great. but
i'll stick to FreeSwitch not Asterisk.



On Fri, Jun 17, 2011 at 10:57 PM, Michelle Konzack <
linux4michelle at tamay-dogan.net> wrote:

> Hello Nandy Dagondon,
>
> Am 2011-06-17 14:07:54, hacktest Du folgendes herunter:
> > hello guys!
> >
> > i'm trying to setup a small exchange w/ Internet service - of course,
> using
> > FS as the exchange. DSLAMs connect to POTS via 2-wire splitters and that
> > would require using FXS gateway (as what Rob mentioned).
> >
> > i'm trying to find out if there's a way we can omit the FXS - so it's
> > FS-to-DSLAM direct via IP.  i'm not sure if this setup is now possible on
> > newer IP DSLAMs.
>
> WHY not give your customers a ADSL/VoIP Gateway where they  connect  the
> telephone tothe router?  And then they can connect directly to Asterisk.
>
> Thanks, Greetings and nice Day/Evening
>    Michelle Konzack
>
> --
> ##################### Debian GNU/Linux Consultant ######################
>   Development of Intranet and Embedded Systems with Debian GNU/Linux
>
> itsystems at tdnet France            itsystems at tdnet
> Owner Michelle Konzack            Owner Michelle Konzack
>
> Apt. 917 (homeoffice)             Gewerbe Stra?e 3
> 50, rue de Soultz                 77694 Kehl/Germany
> 67100 Strasbourg/France           Tel: +49-177-9351947  mobil
> Tel: +33-6-61925193 mobil         Tel: +49-176-86004575 office
>
> <http://www.itsystems.tamay-dogan.net/>  <http://www.flexray4linux.org/>
> <http://www.debian.tamay-dogan.net/>         <http://www.can4linux.org/>
>
> Jabber linux4michelle at jabber.ccc.de
> ICQ    #328449886
>
> Linux-User #280138 with the Linux Counter, http://counter.li.org/
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From linux4michelle at tamay-dogan.net  Sat Jun 18 02:49:13 2011
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Sat, 18 Jun 2011 00:49:13 +0200
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <BANLkTi=GKEKqmSQzKr9KSsXLGfRHUOut4g@mail.gmail.com>
References: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
	<201106162131.04859.justlikeef@gmail.com>
	<BANLkTi=5LJDDxzY2d3nirUnQGpOqoBC45Q@mail.gmail.com>
	<BANLkTimY5oG_1-Tpk5eBKKC1xX8fXk3AoQ@mail.gmail.com>
	<20110617145714.GX4017@michelle1>
	<BANLkTi=GKEKqmSQzKr9KSsXLGfRHUOut4g@mail.gmail.com>
Message-ID: <20110617224913.GC4017@michelle1>

Hello Nandy Dagondon,

Am 2011-06-18 05:54:23, hacktest Du folgendes herunter:
> actually this is what we need if the subscriber needs Internet connection.
> any suggested brand/model of ADSL/VoIP gateway? that would be great. but
> i'll stick to FreeSwitch not Asterisk.

There are MANY DSL/VoIP Router.  We in Germany use mainly AVM but I have
also products from Matrix (India) and some chinese manufacturers.

Price is in the range between 40 and 300 US$ depending on the  requested
features.

Thanks, Greetings and nice Day/Evening
    Michelle Konzack

-- 
##################### Debian GNU/Linux Consultant ######################
   Development of Intranet and Embedded Systems with Debian GNU/Linux

itsystems at tdnet France            itsystems at tdnet
Owner Michelle Konzack            Owner Michelle Konzack

Apt. 917 (homeoffice)             Gewerbe Stra?e 3
50, rue de Soultz                 77694 Kehl/Germany
67100 Strasbourg/France           Tel: +49-177-9351947  mobil
Tel: +33-6-61925193 mobil         Tel: +49-176-86004575 office

<http://www.itsystems.tamay-dogan.net/>  <http://www.flexray4linux.org/>
<http://www.debian.tamay-dogan.net/>         <http://www.can4linux.org/>

Jabber linux4michelle at jabber.ccc.de
ICQ    #328449886

Linux-User #280138 with the Linux Counter, http://counter.li.org/
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From msc at freeswitch.org  Sat Jun 18 03:07:17 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 17 Jun 2011 16:07:17 -0700
Subject: [Freeswitch-users] Generating ringback if 180 arrives after 183
In-Reply-To: <7d1e7eba-6248-44e5-bfee-3723369e1784@mx.voismart.com>
References: <BANLkTinSBBC3icB5UMYiA5O4jCetegMxLA@mail.gmail.com>
	<7d1e7eba-6248-44e5-bfee-3723369e1784@mx.voismart.com>
Message-ID: <BANLkTinHPLci2qA65YYHivvG3fk12K8bJA@mail.gmail.com>

On Fri, Jun 17, 2011 at 1:43 PM, Matteo <mbrancaleoni at voismart.it> wrote:

> Hi,
>
> ----- Messaggio originale -----
> >
> > Any chance you can find the person who designed their SIP stack and
> > hit him with a ClueBat?
>
> why? I think that can be correct, even on sip implementors someone
> expresses
> that if we receive 180 after 183 local ringing must be generated.
> And this is "more right" if before the 180 an update which tells us
> to stop the media is sent, like in this case.
>
> or I'm missing something?
>

I was just being glib. The simple fact of the matter is that if they are
going to send a 183/sdp and then a 180 w/o sdp then you're simply going to
have to figure out how to dance around it, or you're going to need to ignore
that early media and just provide your own ringback to the calling party.

-MC
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From msc at freeswitch.org  Sat Jun 18 04:11:07 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 17 Jun 2011 17:11:07 -0700
Subject: [Freeswitch-users] Help With ClueCon 2011
Message-ID: <BANLkTiniLshnnyD_6tgpTpVWQMFrCF7jGw@mail.gmail.com>

Hey gang!

I am looking to get a few notable names to ClueCon this year and am looking
for businesses who are willing to sponsor by purchasing plane tickets or
hotel rooms. If your business is in a position to assist then please contact
me off list and I will give you the details.

Also, if you haven't already signed up for ClueCon then maybe you'd like to
call in via our new RTMP phone <http://conference.freeswitch.org/cc2/>!
Brian would love to hear from you. :)

Thanks,
Michael
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From msc at freeswitch.org  Sat Jun 18 07:14:33 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 17 Jun 2011 20:14:33 -0700
Subject: [Freeswitch-users] FreeSWITCH Cookbook Recipe List
Message-ID: <BANLkTima7xQPi2JYe908CfyZaYmOMsw=aQ@mail.gmail.com>

Hello all!

Some have inquired recently about what's needed to complete the FS cookbook.
I've uploaded the most recent list of recipes that we have identified. Some
are already written but many are yet to be done. Check out this page:

http://wiki.freeswitch.org/wiki/Cookbook#Information_for_prospective_authors

Contact me if you have any questions about helping with any of these
recipes.

Thanks,
Michael
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From msc at freeswitch.org  Sat Jun 18 08:01:11 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 17 Jun 2011 21:01:11 -0700
Subject: [Freeswitch-users] using keypress to stop a recording.
In-Reply-To: <4DFA7B2D.5040005@499x.com>
References: <4DFA517C.6010109@499x.com>
	<BANLkTimScEmhFtq4sW44CU5LPCifpcEoiQ@mail.gmail.com>
	<4DFA75C1.3050008@499x.com>
	<BANLkTikY45-Pys5+9rY48nT_+Zx+Vxfokg@mail.gmail.com>
	<4DFA7B2D.5040005@499x.com>
Message-ID: <BANLkTim3KJn4QXF+1t0nKtBtMfbn8fJbMA@mail.gmail.com>

On Thu, Jun 16, 2011 at 2:52 PM, Wes <wes-fs at 499x.com> wrote:

> **
> ok, tried that, but it didn't change the behavior  In fact, I removed both
> the onInput function, and the call to session:setVariable, and the recording
> is still halted by *any* keypress on the keypad.  So this must be the
> default behavior of how to stop the recording...
>
> Can anyone confirm this?
>
> I can. It does indeed always stop the recording. Not sure why. However,
there is a simple workaround. Instead of using setInputCallback just do
this:

session:setVariable('playback_terminators','#');
session:execute('record','/tmp/foo.wav');

You would, though, need to setInputCallback while you are playing the
followup message to the caller. BTW, if you have the FS book I did an
example of this, in Lua, where the caller records something then he can
listen, accept, or re-record. Check out chapter 7. FYI, download the code
samples because there's a bug in the text. (Missing the session:ready()
check on one or more while loops, IIRC.)

-MC


> Thanks.
>
>
>
> On 6/16/2011 4:32 PM, Michael Collins wrote:
>
> It's just a channel variable. Set it prior to calling the Lua script or use
> session:setVariable("playback_terminators","#")
>
>  -MC
>
> On Thu, Jun 16, 2011 at 2:29 PM, Wes <wes-fs at 499x.com> wrote:
>
>>  I'm writing a lua script, that link you mention doesn't seem like
>> something I can do in LUA...
>>
>> the problem is in this function:
>>
>>
>> function onInput(s, type, obj)
>>   if (type == "dtmf" and obj['digit'] == '#') then
>>       return "break";
>>   end
>> end
>>
>>  which seems to break on any keypress, so the check for # doesn't work as
>> expected.
>>
>> is there a different way to do this with a lua script?  Ideally, I'd like
>> to do more than just break, I'd like the user to be able to review the
>> message more than once, and then finally hit another key to submit it for
>> real.  Like a voicemail system, actually.
>>
>>
>> On 6/16/2011 3:55 PM, Michael Collins wrote:
>>
>>  How about setting this only to #?
>> http://wiki.freeswitch.org/wiki/Channel_Variables#playback_terminators
>>
>>  -MC
>>
>> On Thu, Jun 16, 2011 at 11:54 AM, Wes <wes-fs at 499x.com> wrote:
>>
>>> I'm using the following script to record, and I"m hoping to take a user
>>> keypress to trigger the end of the recording and then play it back:
>>>
>>> (the problem is that it stops on ANY keypress, while it seems like it
>>> should only break on a "#" keypress)
>>>
>>> I found this example at:
>>> http://wiki.freeswitch.org/wiki/Mod_lua#session:recordFile
>>>
>>> local numberToCall = 1234
>>> local session =
>>> freeswitch.Session("sofia/xxx.xxx.xxx.xxx/"..numberToCall);
>>> session:set_tts_parms("flite", "kal");
>>> session:speak("Thank you for using my recording service, press the pound
>>> key to stop the recording."  );
>>>
>>> function onInput(s, type, obj)
>>>   if (type == "dtmf" and obj['digit'] == '#') then
>>>       return "break";
>>>   end
>>> end
>>>
>>> session:setInputCallback("onInput", "");
>>> session:recordFile("/tmp/luatest.wav");
>>> freeswitch.consoleLog("info", "recording stopped by user keypress \n");
>>> session:speak("your voicehas been recorded, i will play it for you now");
>>> session:streamFile("/tmp/luatest.wav");
>>> session:speak("that's it, goodbye.");
>>> freeswitch.consoleLog("info", "hanging up... \n");
>>> session:hangup();
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From yehavi.bourvine at gmail.com  Sat Jun 18 09:22:39 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sat, 18 Jun 2011 08:22:39 +0300
Subject: [Freeswitch-users] FreeSWITCH Cookbook Need: Fax Examples
In-Reply-To: <BANLkTimzmAwyedYhBFuSy1TMgaQ45tZ-Xg@mail.gmail.com>
References: <BANLkTimzmAwyedYhBFuSy1TMgaQ45tZ-Xg@mail.gmail.com>
Message-ID: <BANLkTikEu27-W5qb8T5pzskqKmYuxoGieA@mail.gmail.com>

Hello Mike,

  We can aid with receiving faxes (although our working example is quite
similar to the one on the WIKI) and T.38 configs.

In about a month we'll be able to provide also email2fax examples (did not
start working on it yet).

                    Regards, __Yehavi:

 2011/6/12 Michael Collins <msc at freeswitch.org>

> Hey all,
>
> I am working on the FSCB and I could use some of your real-world fax
> examples. I'm looking for anything like these:
>
> Inbound fax to email/PDF
> Outbound fax examples (scripts, etc. - whatever you do to make it easier
> for your users to send faxes thru FS)
> T.38 config examples and reports on symptoms/issues you had to overcome to
> make it work
>
> I will gladly accept any information you have, even if it's just raw dumps
> of scripts, XML configs, etc. Shoot them to me off-list.
>
> Thanks!
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mbrancaleoni at voismart.it  Sat Jun 18 11:50:03 2011
From: mbrancaleoni at voismart.it (Matteo)
Date: Sat, 18 Jun 2011 09:50:03 +0200 (CEST)
Subject: [Freeswitch-users] Generating ringback if 180 arrives after 183
In-Reply-To: <BANLkTinHPLci2qA65YYHivvG3fk12K8bJA@mail.gmail.com>
Message-ID: <0503e1d1-37d5-4087-b198-880cf316702e@mx.voismart.com>

> ----- Messaggio originale -----
> I was just being glib. The simple fact of the matter is that if they
> are going to send a 183/sdp and then a 180 w/o sdp then you're
> simply going to have to figure out how to dance around it, or you're
> going to need to ignore that early media and just provide your own
> ringback to the calling party.

Unfortunately I cannot ignore early media since contains the announcement
about the call being transferred.

But I can use the trigger on the update method which stops the media
to do something.

the question is, there's some point into the code to look at in order
to understand how to stop the audio bridge without killing the channels
in order to start the ringback.
I may try coding something... just need some pointers :)

regards,
mat


From freeswitch at peely.com  Sat Jun 18 13:22:39 2011
From: freeswitch at peely.com (peely)
Date: Sat, 18 Jun 2011 02:22:39 -0700 (PDT)
Subject: [Freeswitch-users] Can't make mod_rtmp
In-Reply-To: <BANLkTimx1MnAhfhHVmVds3qYpqNpaC9JKQ@mail.gmail.com>
References: <1308334343913-6488229.post@n2.nabble.com>
	<BANLkTimx1MnAhfhHVmVds3qYpqNpaC9JKQ@mail.gmail.com>
Message-ID: <1308388959641-6490318.post@n2.nabble.com>

That worked, thanks!

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229p6490318.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From sascha.daniels at amooma.de  Sat Jun 18 16:14:50 2011
From: sascha.daniels at amooma.de (Sascha Daniels)
Date: Sat, 18 Jun 2011 14:14:50 +0200
Subject: [Freeswitch-users] Hooks for own call logs
In-Reply-To: <BANLkTim3mx89_p2+KmXk0-iLaBAXQC1EGA@mail.gmail.com>
References: <BANLkTinz0cptgEt3JQ8Eq2FDuNbgsfDdLw@mail.gmail.com>
	<BANLkTim3mx89_p2+KmXk0-iLaBAXQC1EGA@mail.gmail.com>
Message-ID: <4DFC96BA.1080306@amooma.de>

Hi.

Am 17.06.2011 03:43, schrieb Michael Collins:
> It depends on what you mean by "canceled" - can you pastebin a call
> log of a canceled call and give us an idea what should be happening?
>

I would like to make an http call to my application after the originator
canceled the call.

Here ist the cli output.

http://pastebin.com/fCadQTMQ

Somewhere after line 96 the script (or what ever) should be called.

Thanks a lot.

Regards

                    Sascha


-- 
AMOOMA GmbH - Bachstr. 124 - 56566 Neuwied  -->  http://www.amooma.de
Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister Montabaur B14998

B?cher: http://das-asterisk-buch.de - http://ruby-auf-schienen.de



From shouldbeq931 at gmail.com  Sat Jun 18 19:27:22 2011
From: shouldbeq931 at gmail.com (shouldbe q931)
Date: Sat, 18 Jun 2011 17:27:22 +0200
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <54DA215E-D28E-4CCB-9B86-A8DAF1E6BCBA@ipeva.fr>
References: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
	<201106162131.04859.justlikeef@gmail.com>
	<BANLkTi=5LJDDxzY2d3nirUnQGpOqoBC45Q@mail.gmail.com>
	<BANLkTimY5oG_1-Tpk5eBKKC1xX8fXk3AoQ@mail.gmail.com>
	<C201E991-C3A1-437E-BC22-C82FD7360FC1@ipeva.fr>
	<BANLkTin5YSNPPkQfLNuht=yTxmgKeRWAVw@mail.gmail.com>
	<45FF3438-C239-48D9-A57B-F0A49166900E@ipeva.fr>
	<BANLkTin3apQVJLXh-novL0deaq+QM0TuTQ@mail.gmail.com>
	<0013F8A2-8D82-470C-B8D8-45C22BDB3A00@ipeva.fr>
	<BANLkTinLho7z+XgBHGjUJY7+vyaWV60NMA@mail.gmail.com>
	<BANLkTimSqd+XqBQMh5HtmxCfKx+1hZKvvg@mail.gmail.com>
	<54DA215E-D28E-4CCB-9B86-A8DAF1E6BCBA@ipeva.fr>
Message-ID: <BANLkTikut2zrC+_bDTmZbfDLVdg9MnB0rA@mail.gmail.com>

On Fri, Jun 17, 2011 at 10:16 PM, David Ponzone <david.ponzone at ipeva.fr> wrote:
> Draytek gear is crap like hell.
> Stay away from that.
> The support is crap too.
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
granted I have no experience with their "telco" side equipment, but
I've not had any problems with their DSL and Ethernet routers, I've
used them for several commercial deployments where the client didn't
want the expense of Cisco...

the other side of the coin being that I've not needed much in the way
of support, but when I have it's been from their main UK reseller
which has been fine.


From tculjaga at gmail.com  Sat Jun 18 20:00:49 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Sat, 18 Jun 2011 18:00:49 +0200
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <BANLkTikut2zrC+_bDTmZbfDLVdg9MnB0rA@mail.gmail.com>
References: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
	<201106162131.04859.justlikeef@gmail.com>
	<BANLkTi=5LJDDxzY2d3nirUnQGpOqoBC45Q@mail.gmail.com>
	<BANLkTimY5oG_1-Tpk5eBKKC1xX8fXk3AoQ@mail.gmail.com>
	<C201E991-C3A1-437E-BC22-C82FD7360FC1@ipeva.fr>
	<BANLkTin5YSNPPkQfLNuht=yTxmgKeRWAVw@mail.gmail.com>
	<45FF3438-C239-48D9-A57B-F0A49166900E@ipeva.fr>
	<BANLkTin3apQVJLXh-novL0deaq+QM0TuTQ@mail.gmail.com>
	<0013F8A2-8D82-470C-B8D8-45C22BDB3A00@ipeva.fr>
	<BANLkTinLho7z+XgBHGjUJY7+vyaWV60NMA@mail.gmail.com>
	<BANLkTimSqd+XqBQMh5HtmxCfKx+1hZKvvg@mail.gmail.com>
	<54DA215E-D28E-4CCB-9B86-A8DAF1E6BCBA@ipeva.fr>
	<BANLkTikut2zrC+_bDTmZbfDLVdg9MnB0rA@mail.gmail.com>
Message-ID: <BANLkTinC0wVQ5NjwPzdFowbL6i4G3sjY_w@mail.gmail.com>

well, my guess is that he wants a SIP based MSAN with additional FXS ports
on ADSL/VDSL CPEs.

something like this:

---voice---- FS -----SIP---- MSAN ---DSL line---SPL -----low
band---------POTS line (FXS)
                              |                  |
---data-----------------------+                  +-------high
band--------ADSL/VDSL2 Modem W FXS ports (SIP)


anyhow, here everything registers to a FS (MSAN POTS line, FXS ports on the
ADSL modem). FS is in charge to do voice routing, while the data traffic
goes via VLANs to a data network in order to achieve service separation and
an decent level of QoS.
Port density is not an issue here as small density is mainly used for
shopping centers, not residential.


anyhow, i've been working with Huawei MSANs, ZTE MSANs, Zyxel, Siemens. All
of them support H.248 by default... even the micro ones (16 ports from
Zyxel). For the SIP protocol you need to ask a different firmware but its
feasible... they do have it.

in terms of density:

Huawei UA5000 - 660 ports (combo xDSL + POTS), it spoortst ISDN BRI/PRI ,
G.SHDSL ... all sort of things but we are using it for ADSL/VDSL + POTS
Huawei MA500 - 48 ports (same as above)
ZTE ZXDSL 9806H - 64 ports (best buy) ... supports AC or DC power supply in
the same time.
Zyxel - 48 ports ( don't try this at home :P )
Siemens - 48 ( T.30 fax not working well)

T.


On Sat, Jun 18, 2011 at 5:27 PM, shouldbe q931 <shouldbeq931 at gmail.com>wrote:

> On Fri, Jun 17, 2011 at 10:16 PM, David Ponzone <david.ponzone at ipeva.fr>
> wrote:
> > Draytek gear is crap like hell.
> > Stay away from that.
> > The support is crap too.
> > David Ponzone  Direction Technique
> > email: david.ponzone at ipeva.fr
> > tel:      01 74 03 18 97
> > gsm:   06 66 98 76 34
> > Service Client IPeva
> > tel:      0811 46 26 26
> > www.ipeva.fr  -   www.ipeva-studio.com
> > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> > l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion
> > non autoris?e est interdite. Tout message ?lectronique est susceptible
> > d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
> s'il
> > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> >
> >
> granted I have no experience with their "telco" side equipment, but
> I've not had any problems with their DSL and Ethernet routers, I've
> used them for several commercial deployments where the client didn't
> want the expense of Cisco...
>
> the other side of the coin being that I've not needed much in the way
> of support, but when I have it's been from their main UK reseller
> which has been fine.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tculjaga at gmail.com  Sat Jun 18 20:09:01 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Sat, 18 Jun 2011 18:09:01 +0200
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <BANLkTinC0wVQ5NjwPzdFowbL6i4G3sjY_w@mail.gmail.com>
References: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
	<201106162131.04859.justlikeef@gmail.com>
	<BANLkTi=5LJDDxzY2d3nirUnQGpOqoBC45Q@mail.gmail.com>
	<BANLkTimY5oG_1-Tpk5eBKKC1xX8fXk3AoQ@mail.gmail.com>
	<C201E991-C3A1-437E-BC22-C82FD7360FC1@ipeva.fr>
	<BANLkTin5YSNPPkQfLNuht=yTxmgKeRWAVw@mail.gmail.com>
	<45FF3438-C239-48D9-A57B-F0A49166900E@ipeva.fr>
	<BANLkTin3apQVJLXh-novL0deaq+QM0TuTQ@mail.gmail.com>
	<0013F8A2-8D82-470C-B8D8-45C22BDB3A00@ipeva.fr>
	<BANLkTinLho7z+XgBHGjUJY7+vyaWV60NMA@mail.gmail.com>
	<BANLkTimSqd+XqBQMh5HtmxCfKx+1hZKvvg@mail.gmail.com>
	<54DA215E-D28E-4CCB-9B86-A8DAF1E6BCBA@ipeva.fr>
	<BANLkTikut2zrC+_bDTmZbfDLVdg9MnB0rA@mail.gmail.com>
	<BANLkTinC0wVQ5NjwPzdFowbL6i4G3sjY_w@mail.gmail.com>
Message-ID: <BANLkTim6ZZ65Z_VL-ePODy4PsQmEJyBtjg@mail.gmail.com>

As for ADSL/VDSL CPE with FXS ports ... as someone said ... $40 - $300
depending of what is your setup. Whether you need VLAN tagging on the LAN
side, multiple PVCs, VLAN rewriting, IGMP, multiple bridges .. etc
All sort of additional features brings you 1 step towards that $300 mark :=)

my advice to you is to make it simple ... don't complicate things, don't ask
too much from  vendor... Initially you need to present him/them with your
network topology and the services you need to run on that CPE. They will
come back to you with the information whether they can or cannot do it ...
once they accept it ... make them work on the setup... just provide them the
lines.

good luck.

T.


On Sat, Jun 18, 2011 at 6:00 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> well, my guess is that he wants a SIP based MSAN with additional FXS ports
> on ADSL/VDSL CPEs.
>
> something like this:
>
> ---voice---- FS -----SIP---- MSAN ---DSL line---SPL -----low
> band---------POTS line (FXS)
>                               |                  |
> ---data-----------------------+                  +-------high
> band--------ADSL/VDSL2 Modem W FXS ports (SIP)
>
>
> anyhow, here everything registers to a FS (MSAN POTS line, FXS ports on the
> ADSL modem). FS is in charge to do voice routing, while the data traffic
> goes via VLANs to a data network in order to achieve service separation and
> an decent level of QoS.
> Port density is not an issue here as small density is mainly used for
> shopping centers, not residential.
>
>
> anyhow, i've been working with Huawei MSANs, ZTE MSANs, Zyxel, Siemens. All
> of them support H.248 by default... even the micro ones (16 ports from
> Zyxel). For the SIP protocol you need to ask a different firmware but its
> feasible... they do have it.
>
> in terms of density:
>
> Huawei UA5000 - 660 ports (combo xDSL + POTS), it spoortst ISDN BRI/PRI ,
> G.SHDSL ... all sort of things but we are using it for ADSL/VDSL + POTS
> Huawei MA500 - 48 ports (same as above)
> ZTE ZXDSL 9806H - 64 ports (best buy) ... supports AC or DC power supply in
> the same time.
> Zyxel - 48 ports ( don't try this at home :P )
> Siemens - 48 ( T.30 fax not working well)
>
> T.
>
>
>
> On Sat, Jun 18, 2011 at 5:27 PM, shouldbe q931 <shouldbeq931 at gmail.com>wrote:
>
>> On Fri, Jun 17, 2011 at 10:16 PM, David Ponzone <david.ponzone at ipeva.fr>
>> wrote:
>> > Draytek gear is crap like hell.
>> > Stay away from that.
>> > The support is crap too.
>> > David Ponzone  Direction Technique
>> > email: david.ponzone at ipeva.fr
>> > tel:      01 74 03 18 97
>> > gsm:   06 66 98 76 34
>> > Service Client IPeva
>> > tel:      0811 46 26 26
>> > www.ipeva.fr  -   www.ipeva-studio.com
>> > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> > l'intention exclusive de ses destinataires. Toute utilisation ou
>> diffusion
>> > non autoris?e est interdite. Tout message ?lectronique est susceptible
>> > d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
>> s'il
>> > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>> >
>> >
>> granted I have no experience with their "telco" side equipment, but
>> I've not had any problems with their DSL and Ethernet routers, I've
>> used them for several commercial deployments where the client didn't
>> want the expense of Cisco...
>>
>> the other side of the coin being that I've not needed much in the way
>> of support, but when I have it's been from their main UK reseller
>> which has been fine.
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From david.ponzone at ipeva.fr  Sat Jun 18 20:14:24 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sat, 18 Jun 2011 18:14:24 +0200
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <BANLkTikut2zrC+_bDTmZbfDLVdg9MnB0rA@mail.gmail.com>
References: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
	<201106162131.04859.justlikeef@gmail.com>
	<BANLkTi=5LJDDxzY2d3nirUnQGpOqoBC45Q@mail.gmail.com>
	<BANLkTimY5oG_1-Tpk5eBKKC1xX8fXk3AoQ@mail.gmail.com>
	<C201E991-C3A1-437E-BC22-C82FD7360FC1@ipeva.fr>
	<BANLkTin5YSNPPkQfLNuht=yTxmgKeRWAVw@mail.gmail.com>
	<45FF3438-C239-48D9-A57B-F0A49166900E@ipeva.fr>
	<BANLkTin3apQVJLXh-novL0deaq+QM0TuTQ@mail.gmail.com>
	<0013F8A2-8D82-470C-B8D8-45C22BDB3A00@ipeva.fr>
	<BANLkTinLho7z+XgBHGjUJY7+vyaWV60NMA@mail.gmail.com>
	<BANLkTimSqd+XqBQMh5HtmxCfKx+1hZKvvg@mail.gmail.com>
	<54DA215E-D28E-4CCB-9B86-A8DAF1E6BCBA@ipeva.fr>
	<BANLkTikut2zrC+_bDTmZbfDLVdg9MnB0rA@mail.gmail.com>
Message-ID: <BD288582-1888-4665-BA25-5676EDFC9693@ipeva.fr>

I found bugs in the NAT layer of the Vigor 2820 that I prefer not to tell you about....

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

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

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




Le 18/06/2011 ? 17:27, shouldbe q931 a ?crit :

> On Fri, Jun 17, 2011 at 10:16 PM, David Ponzone <david.ponzone at ipeva.fr> wrote:
>> Draytek gear is crap like hell.
>> Stay away from that.
>> The support is crap too.
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>> 
>> 
> granted I have no experience with their "telco" side equipment, but
> I've not had any problems with their DSL and Ethernet routers, I've
> used them for several commercial deployments where the client didn't
> want the expense of Cisco...
> 
> the other side of the coin being that I've not needed much in the way
> of support, but when I have it's been from their main UK reseller
> which has been fine.
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From shouldbeq931 at gmail.com  Sat Jun 18 20:31:12 2011
From: shouldbeq931 at gmail.com (shouldbe q931)
Date: Sat, 18 Jun 2011 18:31:12 +0200
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <BD288582-1888-4665-BA25-5676EDFC9693@ipeva.fr>
References: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
	<201106162131.04859.justlikeef@gmail.com>
	<BANLkTi=5LJDDxzY2d3nirUnQGpOqoBC45Q@mail.gmail.com>
	<BANLkTimY5oG_1-Tpk5eBKKC1xX8fXk3AoQ@mail.gmail.com>
	<C201E991-C3A1-437E-BC22-C82FD7360FC1@ipeva.fr>
	<BANLkTin5YSNPPkQfLNuht=yTxmgKeRWAVw@mail.gmail.com>
	<45FF3438-C239-48D9-A57B-F0A49166900E@ipeva.fr>
	<BANLkTin3apQVJLXh-novL0deaq+QM0TuTQ@mail.gmail.com>
	<0013F8A2-8D82-470C-B8D8-45C22BDB3A00@ipeva.fr>
	<BANLkTinLho7z+XgBHGjUJY7+vyaWV60NMA@mail.gmail.com>
	<BANLkTimSqd+XqBQMh5HtmxCfKx+1hZKvvg@mail.gmail.com>
	<54DA215E-D28E-4CCB-9B86-A8DAF1E6BCBA@ipeva.fr>
	<BANLkTikut2zrC+_bDTmZbfDLVdg9MnB0rA@mail.gmail.com>
	<BD288582-1888-4665-BA25-5676EDFC9693@ipeva.fr>
Message-ID: <BANLkTi=WTsE48a-mbSAs17JDJymT=H0U3w@mail.gmail.com>

2011/6/18 David Ponzone <david.ponzone at ipeva.fr>:
> I found bugs in the NAT layer of the Vigor 2820 that I prefer not to tell
> you about....

Is this a security issue, or a breaks protocol x issue ?


From henk at oegema.com  Sat Jun 18 21:27:33 2011
From: henk at oegema.com (Henk Oegema)
Date: Sat, 18 Jun 2011 19:27:33 +0200
Subject: [Freeswitch-users] No more sessions allowed at this time.
Message-ID: <1308418053.2246.4.camel@DELL>

My FS server was working fine until (suddenly) now.
When I want to call a mobile phone (via a gateway) I get busy (?) or
unavailable (?) and get this message:

freeswitch at internal> 2011-06-18 19:23:00.379259 [CRIT] sofia.c:841 No
more sessions allowed at this time.

There is only one user on my system. (myself)

Rgds.
Henk.
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From linux4michelle at tamay-dogan.net  Sun Jun 19 00:31:59 2011
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Sat, 18 Jun 2011 22:31:59 +0200
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <BANLkTi=WTsE48a-mbSAs17JDJymT=H0U3w@mail.gmail.com>
References: <BANLkTin5YSNPPkQfLNuht=yTxmgKeRWAVw@mail.gmail.com>
	<45FF3438-C239-48D9-A57B-F0A49166900E@ipeva.fr>
	<BANLkTin3apQVJLXh-novL0deaq+QM0TuTQ@mail.gmail.com>
	<0013F8A2-8D82-470C-B8D8-45C22BDB3A00@ipeva.fr>
	<BANLkTinLho7z+XgBHGjUJY7+vyaWV60NMA@mail.gmail.com>
	<BANLkTimSqd+XqBQMh5HtmxCfKx+1hZKvvg@mail.gmail.com>
	<54DA215E-D28E-4CCB-9B86-A8DAF1E6BCBA@ipeva.fr>
	<BANLkTikut2zrC+_bDTmZbfDLVdg9MnB0rA@mail.gmail.com>
	<BD288582-1888-4665-BA25-5676EDFC9693@ipeva.fr>
	<BANLkTi=WTsE48a-mbSAs17JDJymT=H0U3w@mail.gmail.com>
Message-ID: <20110618203159.GD4017@michelle1>

Hello shouldbe q931,

Am 2011-06-18 18:31:12, hacktest Du folgendes herunter:
> 2011/6/18 David Ponzone <david.ponzone at ipeva.fr>:
> > I found bugs in the NAT layer of the Vigor 2820 that I prefer not to tell
> > you about....
> 
> Is this a security issue, or a breaks protocol x issue ?

The Vigors can be transformed into a BOTs and Zombies.
Hijacking Vigors from the Internet is easy!

The problem is known since last years and nothing  has  changed.  Not  a
singel firmware upgrade which solv this problem,  even  if  they  claim,
they are working on it.

Thanks, Greetings and nice Day/Evening
    Michelle Konzack

-- 
##################### Debian GNU/Linux Consultant ######################
   Development of Intranet and Embedded Systems with Debian GNU/Linux

itsystems at tdnet France            itsystems at tdnet
Owner Michelle Konzack            Owner Michelle Konzack

Apt. 917 (homeoffice)             Gewerbe Stra?e 3
50, rue de Soultz                 77694 Kehl/Germany
67100 Strasbourg/France           Tel: +49-177-9351947  mobil
Tel: +33-6-61925193 mobil         Tel: +49-176-86004575 office

<http://www.itsystems.tamay-dogan.net/>  <http://www.flexray4linux.org/>
<http://www.debian.tamay-dogan.net/>         <http://www.can4linux.org/>

Jabber linux4michelle at jabber.ccc.de
ICQ    #328449886

Linux-User #280138 with the Linux Counter, http://counter.li.org/
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From shouldbeq931 at gmail.com  Sun Jun 19 01:11:14 2011
From: shouldbeq931 at gmail.com (shouldbe q931)
Date: Sat, 18 Jun 2011 23:11:14 +0200
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <20110618203159.GD4017@michelle1>
References: <BANLkTin5YSNPPkQfLNuht=yTxmgKeRWAVw@mail.gmail.com>
	<45FF3438-C239-48D9-A57B-F0A49166900E@ipeva.fr>
	<BANLkTin3apQVJLXh-novL0deaq+QM0TuTQ@mail.gmail.com>
	<0013F8A2-8D82-470C-B8D8-45C22BDB3A00@ipeva.fr>
	<BANLkTinLho7z+XgBHGjUJY7+vyaWV60NMA@mail.gmail.com>
	<BANLkTimSqd+XqBQMh5HtmxCfKx+1hZKvvg@mail.gmail.com>
	<54DA215E-D28E-4CCB-9B86-A8DAF1E6BCBA@ipeva.fr>
	<BANLkTikut2zrC+_bDTmZbfDLVdg9MnB0rA@mail.gmail.com>
	<BD288582-1888-4665-BA25-5676EDFC9693@ipeva.fr>
	<BANLkTi=WTsE48a-mbSAs17JDJymT=H0U3w@mail.gmail.com>
	<20110618203159.GD4017@michelle1>
Message-ID: <BANLkTinGgn0PKOr=537adNWVjcFhw0anxw@mail.gmail.com>

On Sat, Jun 18, 2011 at 10:31 PM, Michelle Konzack
<linux4michelle at tamay-dogan.net> wrote:
> Hello shouldbe q931,
>
> Am 2011-06-18 18:31:12, hacktest Du folgendes herunter:
>> 2011/6/18 David Ponzone <david.ponzone at ipeva.fr>:
>> > I found bugs in the NAT layer of the Vigor 2820 that I prefer not to tell
>> > you about....
>>
>> Is this a security issue, or a breaks protocol x issue ?
>
> The Vigors can be transformed into a BOTs and Zombies.
> Hijacking Vigors from the Internet is easy!
>
> The problem is known since last years and nothing ?has ?changed. ?Not ?a
> singel firmware upgrade which solv this problem, ?even ?if ?they ?claim,
> they are working on it.
>
> Thanks, Greetings and nice Day/Evening
> ? ?Michelle Konzack

Got a link to something/anything that discusses this ?

Cheers


From david.ponzone at ipeva.fr  Sun Jun 19 11:33:25 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sun, 19 Jun 2011 09:33:25 +0200
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <BANLkTi=WTsE48a-mbSAs17JDJymT=H0U3w@mail.gmail.com>
References: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
	<201106162131.04859.justlikeef@gmail.com>
	<BANLkTi=5LJDDxzY2d3nirUnQGpOqoBC45Q@mail.gmail.com>
	<BANLkTimY5oG_1-Tpk5eBKKC1xX8fXk3AoQ@mail.gmail.com>
	<C201E991-C3A1-437E-BC22-C82FD7360FC1@ipeva.fr>
	<BANLkTin5YSNPPkQfLNuht=yTxmgKeRWAVw@mail.gmail.com>
	<45FF3438-C239-48D9-A57B-F0A49166900E@ipeva.fr>
	<BANLkTin3apQVJLXh-novL0deaq+QM0TuTQ@mail.gmail.com>
	<0013F8A2-8D82-470C-B8D8-45C22BDB3A00@ipeva.fr>
	<BANLkTinLho7z+XgBHGjUJY7+vyaWV60NMA@mail.gmail.com>
	<BANLkTimSqd+XqBQMh5HtmxCfKx+1hZKvvg@mail.gmail.com>
	<54DA215E-D28E-4CCB-9B86-A8DAF1E6BCBA@ipeva.fr>
	<BANLkTikut2zrC+_bDTmZbfDLVdg9MnB0rA@mail.gmail.com>
	<BD288582-1888-4665-BA25-5676EDFC9693@ipeva.fr>
	<BANLkTi=WTsE48a-mbSAs17JDJymT=H0U3w@mail.gmail.com>
Message-ID: <CAEE7CDA-19AB-4190-99E8-822282B5ACEA@ipeva.fr>

Let's say a weird way to do NAT.
To make it short, the algorithm to compute the external source port of a connection was statically computed from source IP and source port (no randomization of any kind).
So basically, if you had a packet coming from internal IP:port and one coming from internal IP+1:port-1, both were using the same outside port....
You can imagine the result of that with RTP and its systematic RTCP flow using the same port+1.
We had no to number the IP Phones using consecutive IP addresses...

The issue is I had trouble to convince the support that was very wrong.

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

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

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




Le 18/06/2011 ? 18:31, shouldbe q931 a ?crit :

> 2011/6/18 David Ponzone <david.ponzone at ipeva.fr>:
>> I found bugs in the NAT layer of the Vigor 2820 that I prefer not to tell
>> you about....
> 
> Is this a security issue, or a breaks protocol x issue ?
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brokendash at gmail.com  Sun Jun 19 21:41:49 2011
From: brokendash at gmail.com (broken dash)
Date: Sun, 19 Jun 2011 12:41:49 -0500
Subject: [Freeswitch-users] mod_pocketsphinx
In-Reply-To: <BANLkTi=7uDtTx4xWJr9tTnwMNuEQvxZXEQ@mail.gmail.com>
References: <4C1997A0.1080909@gmail.com>
	<BA0A1EAE-474B-4FE1-8BB3-03BC4B1C3655@freeswitch.org>
	<4C1ACD2A.70906@gmail.com>
	<D7122611-F34A-40B4-B0BE-C95A3D551E09@freeswitch.org>
	<4C1AE301.30405@gmail.com> <4C3D48A5.6030508@todandlorna.com>
	<C4BCDAB3-5371-48D5-810F-1A8F299529F6@freeswitch.org>
	<AANLkTi=aTwdANAkNR1V5HXnnrZD8FDeRVWKJktWYCGFa@mail.gmail.com>
	<AF932101-9B49-43F6-A488-8E1C5897E88F@freeswitch.org>
	<AANLkTimOskco=-=uDGdU8OPTOnrU0LXD7iAQyKFo8-4e@mail.gmail.com>
	<CE8866A5-FAB1-429E-A911-C3EF0625651C@freeswitch.org>
	<BANLkTi=7uDtTx4xWJr9tTnwMNuEQvxZXEQ@mail.gmail.com>
Message-ID: <BANLkTinfuryZ92qzaXiZzAnU5E7tZ4S24g@mail.gmail.com>

Does anyone think this issue is related to this?

http://wiki.debian.org/ReleaseGoals/LAFileRemoval



On Tue, Jun 14, 2011 at 5:56 PM, broken dash <brokendash at gmail.com> wrote:
> Has this been fixed? Perhaps a workaround or something... It will
> build on Debian Squeeze, but it damn sure doesn't work. :-)
>
>
> freeswitch at 127.0.0.1@internal> version
> FreeSWITCH Version 1.0.head (git-36f812d 2011-06-14 00-35-18 -0400)
>
> freeswitch at 127.0.0.1@internal> load mod_pocketsphinx
> +OK Reloading XML
> -ERR [module load file routine returned an error]
>
> 2011-06-14 17:46:31.721075 [INFO] switch_time.c:1020 Timezone reloaded
> 530 definitions
> freeswitch at 127.0.0.1@internal> 2011-06-14 17:46:31.721075 [CRIT]
> switch_loadable_module.c:928 Error Loading module
> /usr/local/freeswitch/mod/mod_pocketsphinx.so
> **/usr/local/freeswitch/mod/mod_pocketsphinx.so: undefined symbol:
> ngram_model_get_counts**
>
>
>
> On Wed, Dec 8, 2010 at 9:16 AM, Brian West <brian at freeswitch.org> wrote:
>> Compiles fine on CentOS not sure what your issue is but someone that cares
>> about Ubuntu should probably figure it out and post patches if possible.
>> /b
>> On Dec 8, 2010, at 9:02 AM, Jan Kubr wrote:
>>
>> Were you able to reproduce the problem? Not sure where to look to get rid of
>> this :(
>> Jan
>>
>> On Mon, Dec 6, 2010 at 8:19 PM, Brian West?<brian at freeswitch.org>?wrote:
>>>
>>> No clue I'll try to compile it again today.
>>> /b
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>


From a.luppi at seletech.com  Mon Jun 20 11:55:12 2011
From: a.luppi at seletech.com (Alessandro)
Date: Mon, 20 Jun 2011 09:55:12 +0200
Subject: [Freeswitch-users] Call Hangup
Message-ID: <4DFEFCE0.9040809@seletech.com>

Hi,

I have an active call from the extension A to the extension B.
If one of the two extension is disconnected from the network without 
deregistration the call isn't hung up by the server. The call should be 
hangup when FS realizes that the extension isn't in the network?
Is there a time out to be set somewhere?

thanks

Regards

Alessandro Luppi

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



From david.ponzone at ipeva.fr  Mon Jun 20 12:43:01 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 20 Jun 2011 10:43:01 +0200
Subject: [Freeswitch-users] Call Hangup
In-Reply-To: <4DFEFCE0.9040809@seletech.com>
References: <4DFEFCE0.9040809@seletech.com>
Message-ID: <B8D314BD-A38C-4B6D-AC95-84643B433E9C@ipeva.fr>

Alessandro,

there are 2 ways to do that:
Session timers: with this, FS will send a RE-INVITE to phones every X minutes, and if the phone does not answer, it's considerer dead, so it is unregistered and the current calls will be hung up
or
RTP Timeout: this will detect that media is missing, and will hang up the call

Session timers has to be enabled on the SIP Profile with:
<param name="enable-timer" value="true"/>
<param name="session-timeout" value="120"/>

RTP timeout has to be enabled in the SIP Profile with:
     <param name="rtp-timeout-sec" value="300"/>
(use the value you wish. Also, be careful with this, it can lead to issues with some phones when you put a call on hold/mute or if you use VAD)

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

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

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




Le 20/06/2011 ? 09:55, Alessandro a ?crit :

> Hi,
> 
> I have an active call from the extension A to the extension B.
> If one of the two extension is disconnected from the network without 
> deregistration the call isn't hung up by the server. The call should be 
> hangup when FS realizes that the extension isn't in the network?
> Is there a time out to be set somewhere?
> 
> thanks
> 
> Regards
> 
> Alessandro Luppi
> 
> -- 
> Ing. Alessandro Luppi
> Software development
> Seletech srl
> Via Collodi 8, 20052 Monza (MI) - Italy
> Tel: +39.039.5962000 - Fax: +39.039.9716905
> email: a.luppi at seletech.com - Web: www.seletech.com   or   www.seletech.eu
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From a.luppi at seletech.com  Mon Jun 20 13:28:24 2011
From: a.luppi at seletech.com (Alessandro)
Date: Mon, 20 Jun 2011 11:28:24 +0200
Subject: [Freeswitch-users] Call Hangup
In-Reply-To: <B8D314BD-A38C-4B6D-AC95-84643B433E9C@ipeva.fr>
References: <4DFEFCE0.9040809@seletech.com>
	<B8D314BD-A38C-4B6D-AC95-84643B433E9C@ipeva.fr>
Message-ID: <4DFF12B8.5090008@seletech.com>

Hi,

i set
<param name="enable-timer" value="true"/>
<param name="session-timeout" value="120"/>

on sip profile, but doesn't change. FS doesn't hang up the call after 
one of the two extension disappear from the network.
With wireshark i don't see re-invite during the call.

Regards

Alessandro
  Il 20/06/2011 10:43, David Ponzone ha scritto:
> Alessandro,
>
> there are 2 ways to do that:
> Session timers: with this, FS will send a RE-INVITE to phones every X 
> minutes, and if the phone does not answer, it's considerer dead, so it 
> is unregistered and the current calls will be hung up
> or
> RTP Timeout: this will detect that media is missing, and will hang up 
> the call
>
> Session timers has to be enabled on the SIP Profile with:
> <param name="enable-timer" value="true"/>
> <param name="session-timeout" value="120"/>
>
> RTP timeout has to be enabled in the SIP Profile with:
> <param name="rtp-timeout-sec" value="300"/>
> (use the value you wish. Also, be careful with this, it can lead to 
> issues with some phones when you put a call on hold/mute or if you use 
> VAD)
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service ClientIPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
> diffusion non autoris?e est interdite. Tout message ?lectronique est 
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au 
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
> n'?tes pas destinataire de ce message, merci de le d?truire 
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 20/06/2011 ? 09:55, Alessandro a ?crit :
>
>> Hi,
>>
>> I have an active call from the extension A to the extension B.
>> If one of the two extension is disconnected from the network without
>> deregistration the call isn't hung up by the server. The call should be
>> hangup when FS realizes that the extension isn't in the network?
>> Is there a time out to be set somewhere?
>>
>> thanks
>>
>> Regards
>>
>> Alessandro Luppi
>>
>> -- 
>> Ing. Alessandro Luppi
>> Software development
>> Seletech srl
>> Via Collodi 8, 20052 Monza (MI) - Italy
>> Tel: +39.039.5962000 - Fax: +39.039.9716905
>> email: a.luppi at seletech.com <mailto:a.luppi at seletech.com> - Web: 
>> www.seletech.com <http://www.seletech.com>   or www.seletech.eu 
>> <http://www.seletech.eu>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org 
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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

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From ankitwalia4u at gmail.com  Mon Jun 20 13:36:00 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Mon, 20 Jun 2011 15:06:00 +0530
Subject: [Freeswitch-users] No Voice for default IVR ext 5000
In-Reply-To: <BANLkTin8_p2H5MMZK+U7vC4vPnp7=46atg@mail.gmail.com>
References: <BANLkTi=xZ09f_9HNH6D=c8fpTM72qXbxJA@mail.gmail.com>
	<BANLkTinr17B8y2EcwQ0YAVwAGU3giNXauw@mail.gmail.com>
	<BANLkTim7TeR2WxQOCV+-h=CYF-bQPPMUkg@mail.gmail.com>
	<BANLkTikrkZ80n6R_JwyGq1T3RvbG5RrWQQ@mail.gmail.com>
	<BANLkTin8_p2H5MMZK+U7vC4vPnp7=46atg@mail.gmail.com>
Message-ID: <BANLkTik=E3SDn1jaa8gXOvyEcXjD=M3dXg@mail.gmail.com>

Hello Guru, MC and friends,

Sorry for the delayed reply, I did some changes in my broadband modem, reset
the conf

On Sat, Jun 18, 2011 at 1:58 AM, guru singh <grsingh750 at gmail.com> wrote:

> Hi Ankit,
>
> You could also see if there are SIP ALG settings on the router.
>
> If by broadband router you mean the DSL modem that providers give,
> then you're in a sticky situation. From my experience they are very
> unreliable, something that works would break without any apparent
> reason or even worse at times things would only work intermittently.
> So you just end up racking your brains trying to reproduce and
> diagnose, what I'd call "a false problem". Replacing that would reduce
> some of your troubles.
>
> Regards,
> guru
>
> On Sat, Jun 18, 2011 at 12:35 AM, Michael Collins <msc at freeswitch.org>
> wrote:
> > I would get a pcap of both the signaling and the RTP and analyze w/
> > wireshark. Find out if the RTP is actually leaving FreeSWITCH or not. Of
> > course, you might start with a console debug log and siptrace on
> > pastebin.freeswitch.org so that we can have a looksee at what is
> happening
> > on the call.
> > -MC
> >
> > On Fri, Jun 17, 2011 at 9:07 AM, ankIT WALiA <ankitwalia4u at gmail.com>
> wrote:
> >>
> >> Yes, I am under NAT through my Broadband router.
> >> But, if NAT is the issue, then is it possible for both sides able to
> >> listen each other voices when I am transferring my call to another ext.
> >> which is not IVR
> >> Could someone please suggest what I can do to work with NAT or under
> >> firewall?
> >> Secondly, do you know any other reason for this issue if its not NAT?
> >> Thanks
> >> Ankit
> >>
> >> On Fri, Jun 17, 2011 at 7:05 PM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >>>
> >>> Is there any NAT involved? Usually lack of audio means that a NAT
> router
> >>> or firewall is screwing up the RTP media which is sent in a separate
> UDP
> >>> stream. Either it's not being forwarded, or it's being told to go to
> the
> >>> wrong place.
> >>>
> >>> -Steve
> >>>
> >>>
> >>> On 17 June 2011 13:13, ankIT WALiA <ankitwalia4u at gmail.com> wrote:
> >>>>
> >>>> Dear all,
> >>>> I have configured an gateway for external profile to receive and make
> >>>> calls.
> >>>> I have added a file in public directory.
> >>>> <include>
> >>>> <extension name="iptel-inbound">
> >>>> <condition field="destination_number"
> >>>> expression="^(MY_IPTEL_USERNAME)$">
> >>>> <action application="set" data="domain_name=$${domain}"/>
> >>>> <action application="transfer" data="ExtNo XML default"/>
> >>>> </condition>
> >>>> </extension>
> >>>> </include>
> >>>> Now, while I am calling to my voip account, I am transferring the
> call.
> >>>> It is getting transferred to the extension also.
> >>>> But, If I use normal ext number, I am able to talk both ways. But when
> I
> >>>> use default ext number 5000 for IVR menu. I could not listen to IVR
> menu.
> >>>> Though, I can see in my logs that the audio file are being played one
> by
> >>>> one, but I cant listen.
> >>>> Do I need some extra configuration to listen to IVR?
> >>>> There may be a gap in my knowledge. Please enlighten.
> >>>> Thanks
> >>>> Ankit
> >>>> --
> >>>> Life is like a rose its upto u feel it as its fragrance or thorns
> >>>>
> >>>> _______________________________________________
> >>>> Join us at ClueCon 2011, Aug 9-11, Chicago
> >>>> http://www.cluecon.com 877-7-4ACLUE
> >>>>
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>
> >>>
> >>> _______________________________________________
> >>> Join us at ClueCon 2011, Aug 9-11, Chicago
> >>> http://www.cluecon.com 877-7-4ACLUE
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >>
> >> --
> >> Life is like a rose its upto u feel it as its fragrance or thorns
> >>
> >> _______________________________________________
> >> Join us at ClueCon 2011, Aug 9-11, Chicago
> >> http://www.cluecon.com 877-7-4ACLUE
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Life is like a rose its upto u feel it as its fragrance or thorns
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From ankitwalia4u at gmail.com  Mon Jun 20 13:45:37 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Mon, 20 Jun 2011 15:15:37 +0530
Subject: [Freeswitch-users] No Voice for default IVR ext 5000
In-Reply-To: <BANLkTik=E3SDn1jaa8gXOvyEcXjD=M3dXg@mail.gmail.com>
References: <BANLkTi=xZ09f_9HNH6D=c8fpTM72qXbxJA@mail.gmail.com>
	<BANLkTinr17B8y2EcwQ0YAVwAGU3giNXauw@mail.gmail.com>
	<BANLkTim7TeR2WxQOCV+-h=CYF-bQPPMUkg@mail.gmail.com>
	<BANLkTikrkZ80n6R_JwyGq1T3RvbG5RrWQQ@mail.gmail.com>
	<BANLkTin8_p2H5MMZK+U7vC4vPnp7=46atg@mail.gmail.com>
	<BANLkTik=E3SDn1jaa8gXOvyEcXjD=M3dXg@mail.gmail.com>
Message-ID: <BANLkTi=UxZXgKgjtUPK-uf-x4ikQh_iw5g@mail.gmail.com>

Hello Guru, MC, Steve and friends,

I have pasted my logs on http://pastebin.freeswitch.org/16555

I observed the logs and saw the below line:
2011-06-20 14:57:27.282184 [DEBUG] sofia_glue.c:2594 AUDIO RTP
[sofia/external/waliaankit1 at sip.actionvoip.com:5060] 192.168.1.2 port 23520
-> 77.72.168.99 port 11994 codec: 8 ms: 20

I think it is telling me that the request is from IP address 192.168.1.2
port 23520 to 77.72.168.99 port 11994.Though, I am using NAT at my broadband
modem but my public IP address is also not 77.72.168.99.

I think as told by Steve, the response is going to wrong place.

Guru,
I also saw the SIP ALG setting, SIP is enabled.

Please suggest what I can do in this case. I can try wireshark or pcap, but
isnt it evident from the above log that the response is going to the wrong
place.

Can it be a problem that my FS server, my call request system and my
receiving system is one system only having one network device with IP
address 192.168.1.2 ?

Thanks,
Ankit



On Mon, Jun 20, 2011 at 3:06 PM, ankIT WALiA <ankitwalia4u at gmail.com> wrote:

> Hello Guru, MC and friends,
>
> Sorry for the delayed reply, I did some changes in my broadband modem,
> reset the conf
>
>
> On Sat, Jun 18, 2011 at 1:58 AM, guru singh <grsingh750 at gmail.com> wrote:
>
>> Hi Ankit,
>>
>> You could also see if there are SIP ALG settings on the router.
>>
>> If by broadband router you mean the DSL modem that providers give,
>> then you're in a sticky situation. From my experience they are very
>> unreliable, something that works would break without any apparent
>> reason or even worse at times things would only work intermittently.
>> So you just end up racking your brains trying to reproduce and
>> diagnose, what I'd call "a false problem". Replacing that would reduce
>> some of your troubles.
>>
>> Regards,
>> guru
>>
>> On Sat, Jun 18, 2011 at 12:35 AM, Michael Collins <msc at freeswitch.org>
>> wrote:
>> > I would get a pcap of both the signaling and the RTP and analyze w/
>> > wireshark. Find out if the RTP is actually leaving FreeSWITCH or not. Of
>> > course, you might start with a console debug log and siptrace on
>> > pastebin.freeswitch.org so that we can have a looksee at what is
>> happening
>> > on the call.
>> > -MC
>> >
>> > On Fri, Jun 17, 2011 at 9:07 AM, ankIT WALiA <ankitwalia4u at gmail.com>
>> wrote:
>> >>
>> >> Yes, I am under NAT through my Broadband router.
>> >> But, if NAT is the issue, then is it possible for both sides able to
>> >> listen each other voices when I am transferring my call to another ext.
>> >> which is not IVR
>> >> Could someone please suggest what I can do to work with NAT or under
>> >> firewall?
>> >> Secondly, do you know any other reason for this issue if its not NAT?
>> >> Thanks
>> >> Ankit
>> >>
>> >> On Fri, Jun 17, 2011 at 7:05 PM, Steven Ayre <steveayre at gmail.com>
>> wrote:
>> >>>
>> >>> Is there any NAT involved? Usually lack of audio means that a NAT
>> router
>> >>> or firewall is screwing up the RTP media which is sent in a separate
>> UDP
>> >>> stream. Either it's not being forwarded, or it's being told to go to
>> the
>> >>> wrong place.
>> >>>
>> >>> -Steve
>> >>>
>> >>>
>> >>> On 17 June 2011 13:13, ankIT WALiA <ankitwalia4u at gmail.com> wrote:
>> >>>>
>> >>>> Dear all,
>> >>>> I have configured an gateway for external profile to receive and make
>> >>>> calls.
>> >>>> I have added a file in public directory.
>> >>>> <include>
>> >>>> <extension name="iptel-inbound">
>> >>>> <condition field="destination_number"
>> >>>> expression="^(MY_IPTEL_USERNAME)$">
>> >>>> <action application="set" data="domain_name=$${domain}"/>
>> >>>> <action application="transfer" data="ExtNo XML default"/>
>> >>>> </condition>
>> >>>> </extension>
>> >>>> </include>
>> >>>> Now, while I am calling to my voip account, I am transferring the
>> call.
>> >>>> It is getting transferred to the extension also.
>> >>>> But, If I use normal ext number, I am able to talk both ways. But
>> when I
>> >>>> use default ext number 5000 for IVR menu. I could not listen to IVR
>> menu.
>> >>>> Though, I can see in my logs that the audio file are being played one
>> by
>> >>>> one, but I cant listen.
>> >>>> Do I need some extra configuration to listen to IVR?
>> >>>> There may be a gap in my knowledge. Please enlighten.
>> >>>> Thanks
>> >>>> Ankit
>> >>>> --
>> >>>> Life is like a rose its upto u feel it as its fragrance or thorns
>> >>>>
>> >>>> _______________________________________________
>> >>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> >>>> http://www.cluecon.com 877-7-4ACLUE
>> >>>>
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>>
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> >>> http://www.cluecon.com 877-7-4ACLUE
>> >>>
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>
>> >>
>> >>
>> >> --
>> >> Life is like a rose its upto u feel it as its fragrance or thorns
>> >>
>> >> _______________________________________________
>> >> Join us at ClueCon 2011, Aug 9-11, Chicago
>> >> http://www.cluecon.com 877-7-4ACLUE
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>> > http://www.cluecon.com 877-7-4ACLUE
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Life is like a rose its upto u feel it as its fragrance or thorns
>



-- 
Life is like a rose its upto u feel it as its fragrance or thorns
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From ankitwalia4u at gmail.com  Mon Jun 20 14:03:50 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Mon, 20 Jun 2011 15:33:50 +0530
Subject: [Freeswitch-users] No Voice for default IVR ext 5000
In-Reply-To: <BANLkTi=UxZXgKgjtUPK-uf-x4ikQh_iw5g@mail.gmail.com>
References: <BANLkTi=xZ09f_9HNH6D=c8fpTM72qXbxJA@mail.gmail.com>
	<BANLkTinr17B8y2EcwQ0YAVwAGU3giNXauw@mail.gmail.com>
	<BANLkTim7TeR2WxQOCV+-h=CYF-bQPPMUkg@mail.gmail.com>
	<BANLkTikrkZ80n6R_JwyGq1T3RvbG5RrWQQ@mail.gmail.com>
	<BANLkTin8_p2H5MMZK+U7vC4vPnp7=46atg@mail.gmail.com>
	<BANLkTik=E3SDn1jaa8gXOvyEcXjD=M3dXg@mail.gmail.com>
	<BANLkTi=UxZXgKgjtUPK-uf-x4ikQh_iw5g@mail.gmail.com>
Message-ID: <BANLkTi=ik56oG4Vi5YGbYXwSNGEKSho4OQ@mail.gmail.com>

When I tried with FS Server and my receivig system as same system with one
IP address.
Call Requesting System from other system with second IP address.
It worked as expected, I can listen the IVR menu.

But the logs says
2011-06-20 15:27:53.054612 [DEBUG] sofia_glue.c:2594 AUDIO RTP
[sofia/external/w
aliaankit1 at sip.actionvoip.com:5060] 192.168.1.2 port 24484 -> 194.120.0.106
port
 11772 codec: 0 ms: 20

My Requesting system IP address is correct but my receiving system IP
address is 192.168.1.3.

Is it something related to NAT? Is it something which I dont know about
Networking? Is this a random behavior?

Thanks
Ankit

On Mon, Jun 20, 2011 at 3:15 PM, ankIT WALiA <ankitwalia4u at gmail.com> wrote:

> Hello Guru, MC, Steve and friends,
>
> I have pasted my logs on http://pastebin.freeswitch.org/16555
>
> I observed the logs and saw the below line:
> 2011-06-20 14:57:27.282184 [DEBUG] sofia_glue.c:2594 AUDIO RTP
> [sofia/external/waliaankit1 at sip.actionvoip.com:5060] 192.168.1.2 port
> 23520 -> 77.72.168.99 port 11994 codec: 8 ms: 20
>
> I think it is telling me that the request is from IP address 192.168.1.2
> port 23520 to 77.72.168.99 port 11994.Though, I am using NAT at my broadband
> modem but my public IP address is also not 77.72.168.99.
>
> I think as told by Steve, the response is going to wrong place.
>
> Guru,
> I also saw the SIP ALG setting, SIP is enabled.
>
> Please suggest what I can do in this case. I can try wireshark or pcap, but
> isnt it evident from the above log that the response is going to the wrong
> place.
>
> Can it be a problem that my FS server, my call request system and my
> receiving system is one system only having one network device with IP
> address 192.168.1.2 ?
>
> Thanks,
> Ankit
>
>
>
> On Mon, Jun 20, 2011 at 3:06 PM, ankIT WALiA <ankitwalia4u at gmail.com>wrote:
>
>> Hello Guru, MC and friends,
>>
>> Sorry for the delayed reply, I did some changes in my broadband modem,
>> reset the conf
>>
>>
>> On Sat, Jun 18, 2011 at 1:58 AM, guru singh <grsingh750 at gmail.com> wrote:
>>
>>> Hi Ankit,
>>>
>>> You could also see if there are SIP ALG settings on the router.
>>>
>>> If by broadband router you mean the DSL modem that providers give,
>>> then you're in a sticky situation. From my experience they are very
>>> unreliable, something that works would break without any apparent
>>> reason or even worse at times things would only work intermittently.
>>> So you just end up racking your brains trying to reproduce and
>>> diagnose, what I'd call "a false problem". Replacing that would reduce
>>> some of your troubles.
>>>
>>> Regards,
>>> guru
>>>
>>> On Sat, Jun 18, 2011 at 12:35 AM, Michael Collins <msc at freeswitch.org>
>>> wrote:
>>> > I would get a pcap of both the signaling and the RTP and analyze w/
>>> > wireshark. Find out if the RTP is actually leaving FreeSWITCH or not.
>>> Of
>>> > course, you might start with a console debug log and siptrace on
>>> > pastebin.freeswitch.org so that we can have a looksee at what is
>>> happening
>>> > on the call.
>>> > -MC
>>> >
>>> > On Fri, Jun 17, 2011 at 9:07 AM, ankIT WALiA <ankitwalia4u at gmail.com>
>>> wrote:
>>> >>
>>> >> Yes, I am under NAT through my Broadband router.
>>> >> But, if NAT is the issue, then is it possible for both sides able to
>>> >> listen each other voices when I am transferring my call to another
>>> ext.
>>> >> which is not IVR
>>> >> Could someone please suggest what I can do to work with NAT or under
>>> >> firewall?
>>> >> Secondly, do you know any other reason for this issue if its not NAT?
>>> >> Thanks
>>> >> Ankit
>>> >>
>>> >> On Fri, Jun 17, 2011 at 7:05 PM, Steven Ayre <steveayre at gmail.com>
>>> wrote:
>>> >>>
>>> >>> Is there any NAT involved? Usually lack of audio means that a NAT
>>> router
>>> >>> or firewall is screwing up the RTP media which is sent in a separate
>>> UDP
>>> >>> stream. Either it's not being forwarded, or it's being told to go to
>>> the
>>> >>> wrong place.
>>> >>>
>>> >>> -Steve
>>> >>>
>>> >>>
>>> >>> On 17 June 2011 13:13, ankIT WALiA <ankitwalia4u at gmail.com> wrote:
>>> >>>>
>>> >>>> Dear all,
>>> >>>> I have configured an gateway for external profile to receive and
>>> make
>>> >>>> calls.
>>> >>>> I have added a file in public directory.
>>> >>>> <include>
>>> >>>> <extension name="iptel-inbound">
>>> >>>> <condition field="destination_number"
>>> >>>> expression="^(MY_IPTEL_USERNAME)$">
>>> >>>> <action application="set" data="domain_name=$${domain}"/>
>>> >>>> <action application="transfer" data="ExtNo XML default"/>
>>> >>>> </condition>
>>> >>>> </extension>
>>> >>>> </include>
>>> >>>> Now, while I am calling to my voip account, I am transferring the
>>> call.
>>> >>>> It is getting transferred to the extension also.
>>> >>>> But, If I use normal ext number, I am able to talk both ways. But
>>> when I
>>> >>>> use default ext number 5000 for IVR menu. I could not listen to IVR
>>> menu.
>>> >>>> Though, I can see in my logs that the audio file are being played
>>> one by
>>> >>>> one, but I cant listen.
>>> >>>> Do I need some extra configuration to listen to IVR?
>>> >>>> There may be a gap in my knowledge. Please enlighten.
>>> >>>> Thanks
>>> >>>> Ankit
>>> >>>> --
>>> >>>> Life is like a rose its upto u feel it as its fragrance or thorns
>>> >>>>
>>> >>>> _______________________________________________
>>> >>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> >>>> http://www.cluecon.com 877-7-4ACLUE
>>> >>>>
>>> >>>> FreeSWITCH-users mailing list
>>> >>>> FreeSWITCH-users at lists.freeswitch.org
>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>> http://www.freeswitch.org
>>> >>>>
>>> >>>
>>> >>>
>>> >>> _______________________________________________
>>> >>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> >>> http://www.cluecon.com 877-7-4ACLUE
>>> >>>
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>>
>>> >>
>>> >>
>>> >>
>>> >> --
>>> >> Life is like a rose its upto u feel it as its fragrance or thorns
>>> >>
>>> >> _______________________________________________
>>> >> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> >> http://www.cluecon.com 877-7-4ACLUE
>>> >>
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>> >
>>> > _______________________________________________
>>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>>> > http://www.cluecon.com 877-7-4ACLUE
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Life is like a rose its upto u feel it as its fragrance or thorns
>>
>
>
>
> --
> Life is like a rose its upto u feel it as its fragrance or thorns
>



-- 
Life is like a rose its upto u feel it as its fragrance or thorns
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From fieldpeak at gmail.com  Mon Jun 20 15:11:07 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Mon, 20 Jun 2011 19:11:07 +0800
Subject: [Freeswitch-users] Can FS use external user database
Message-ID: <BANLkTimmQgOUj72p0HYz4JEqJebO9SBE-Q@mail.gmail.com>

Gurus,

Can we configure FS to have it to read registeration users information (e.g.
account and pwd) for authentication from external database replacing default
xml files(under conf\directory\default)?

e.g. we have a database which contains user's account and password and was
used by other application, we would like to have FS to read this exsiting
databse for authentication as well...

Thanks.

Regards,
Charles
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From jgallartm at gmail.com  Mon Jun 20 15:37:12 2011
From: jgallartm at gmail.com (Javier Gallart)
Date: Mon, 20 Jun 2011 13:37:12 +0200
Subject: [Freeswitch-users] bind_digit_action delay
Message-ID: <BANLkTinFcx7Ud=ajEwwffHkc61v8WnTR6w@mail.gmail.com>

Hello list,

I'm using to bind_digit_action application to launch an event when a
particular sequence of keys (##) is pressed. It works fine, the only problem
is that the application is launched 5 seconds after the last key is pressed.
I've tried with a simple playback application and it's just the same:
2011-06-20 07:22:55.641019 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF #:1560
2011-06-20 07:22:56.121025 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF #:1480
2011-06-20 07:22:56.121025 [DEBUG] mod_dptools.c:151 sofia/internal/
34917019888 at 79.170.64.151 Digit match binding
[exec:playback][/var/local/sounds/common/en/last_time]
(...)
2011-06-20 07:22:56.441011 [DEBUG] switch_ivr.c:567 sofia/internal/
34661574758 at 79.170.68.169:5060 Command Execute playback(local_stream://moh)
EXECUTE sofia/internal/34661574758 at 79.170.68.169:5060playback(local_stream://moh)
2011-06-20 07:22:56.441011 [WARNING] mod_local_stream.c:393 Unknown source
moh, trying 'default'
2011-06-20 07:22:56.441011 [ERR] mod_local_stream.c:402 Unknown source
default
(...)
2011-06-20 07:23:01.281067 [DEBUG] switch_ivr.c:567 sofia/internal/
34917019888 at 79.170.64.151 Command Execute
playback(/var/local/sounds/common/en/last_time)
EXECUTE sofia/internal/34917019888 at 79.170.64.151playback(/var/local/sounds/common/en/last_time)

My first question is why does fs tries to play moh? And the second one is if
there is a way to shorten the lapse between the last key and the execution
of the application. I've played a bit with the bind_digit_input_timeout
variable that I've seen in the code, with no success. Does anyone have faces
the same issue?
I'm using version FreeSWITCH Version 1.0.head (git-7768808 2011-06-18
11-52-37 -0500)

Thanks in advance

Javi
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From peder at networkoblivion.com  Mon Jun 20 15:49:49 2011
From: peder at networkoblivion.com (Peder)
Date: Mon, 20 Jun 2011 06:49:49 -0500
Subject: [Freeswitch-users] Can FS use external user database
In-Reply-To: <BANLkTimmQgOUj72p0HYz4JEqJebO9SBE-Q@mail.gmail.com>
References: <BANLkTimmQgOUj72p0HYz4JEqJebO9SBE-Q@mail.gmail.com>
Message-ID: <05fb01cc2f40$297556f0$7c6004d0$@com>

Not directly, but you can use xml_curl to query a web server which queries a
database:

 

http://wiki.freeswitch.org/wiki/Mod_xml_curl

 

http://wiki.freeswitch.org/wiki/Mod_xml_curl#bindings.3D.22directory.22

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
fieldpeak
Sent: Monday, June 20, 2011 6:11 AM
To: FreeSWITCH-users
Cc: 442962866 at 139.com
Subject: [Freeswitch-users] Can FS use external user database

 

Gurus,

Can we configure FS to have it to read registeration users information (e.g.
account and pwd) for authentication from external database replacing default
xml files(under conf\directory\default)? 

e.g. we have a database which contains user's account and password and was
used by other application, we would like to have FS to read this exsiting
databse for authentication as well...

Thanks.

Regards,
Charles

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From nazim.aghabayov at gmail.com  Mon Jun 20 15:51:05 2011
From: nazim.aghabayov at gmail.com (Nazim Aghabayov)
Date: Mon, 20 Jun 2011 16:51:05 +0500
Subject: [Freeswitch-users] Can FS use external user database
In-Reply-To: <BANLkTimmQgOUj72p0HYz4JEqJebO9SBE-Q@mail.gmail.com>
References: <BANLkTimmQgOUj72p0HYz4JEqJebO9SBE-Q@mail.gmail.com>
Message-ID: <4DFF3429.5020905@gmail.com>

Hello Charles,

Mod xml curl with bindings="directory" does that.
Please have a look at http://wiki.freeswitch.org/wiki/Mod_xml_curl

Best Regards,
Nazim

On 06/20/2011 04:11 PM, fieldpeak wrote:
> Gurus,
>
> Can we configure FS to have it to read registeration users information (e.g.
> account and pwd) for authentication from external database replacing default
> xml files(under conf\directory\default)?
>
> e.g. we have a database which contains user's account and password and was
> used by other application, we would like to have FS to read this exsiting
> databse for authentication as well...
>
> Thanks.
>
> Regards,
> Charles
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From fieldpeak at gmail.com  Mon Jun 20 16:21:01 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Mon, 20 Jun 2011 20:21:01 +0800
Subject: [Freeswitch-users] Can FS use external user database
In-Reply-To: <4DFF3429.5020905@gmail.com>
References: <BANLkTimmQgOUj72p0HYz4JEqJebO9SBE-Q@mail.gmail.com>
	<4DFF3429.5020905@gmail.com>
Message-ID: <BANLkTimj5MkmHaeyPff8_1iXgThS3ODzsA@mail.gmail.com>

Hello Peder and Nazim,

Thanks for your valuable information, i will have a try...

Regards,
Charles

2011/6/20 Nazim Aghabayov <nazim.aghabayov at gmail.com>

> Hello Charles,
>
> Mod xml curl with bindings="directory" does that.
> Please have a look at http://wiki.freeswitch.org/wiki/Mod_xml_curl
>
> Best Regards,
> Nazim
>
> On 06/20/2011 04:11 PM, fieldpeak wrote:
> > Gurus,
> >
> > Can we configure FS to have it to read registeration users information
> (e.g.
> > account and pwd) for authentication from external database replacing
> default
> > xml files(under conf\directory\default)?
> >
> > e.g. we have a database which contains user's account and password and
> was
> > used by other application, we would like to have FS to read this exsiting
> > databse for authentication as well...
> >
> > Thanks.
> >
> > Regards,
> > Charles
> >
> >
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mitch.johnson7 at gmail.com  Mon Jun 20 17:24:48 2011
From: mitch.johnson7 at gmail.com (mitch Johnson)
Date: Mon, 20 Jun 2011 09:24:48 -0400
Subject: [Freeswitch-users] FreeSWITCH Cookbook Recipe List
Message-ID: <BANLkTimn9OzKORpuG259rTRFGPwdLbd9FA@mail.gmail.com>

It seems as if you have already identified what's going into the Cookbook.
Is it too late to add a few sections on TLS/SRTP with examples on using
different phone vendors and softphones on smartphones?

I think FreeSWITCH is a powerful solution in the TLS/SRTP area yet to be
realized or tapped into.  Having recipes on how to implement and a great
discussion on how FreeSWITCH can use channels on a single box to do
encrypted calls on the Internet side and unencrypted calls on the LAN side
would be incredible.

Thanks,

Mitch Johnson

>
> ---------- Forwarded message ----------
> From: Michael Collins <msc at freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org,
> freeswitch-dev at lists.freeswitch.org
> Date: Fri, 17 Jun 2011 20:14:33 -0700
> Subject: [Freeswitch-users] FreeSWITCH Cookbook Recipe List
> Hello all!
>
> Some have inquired recently about what's needed to complete the FS
> cookbook. I've uploaded the most recent list of recipes that we have
> identified. Some are already written but many are yet to be done. Check out
> this page:
>
>
> http://wiki.freeswitch.org/wiki/Cookbook#Information_for_prospective_authors
>
> Contact me if you have any questions about helping with any of these
> recipes.
>
> Thanks,
> Michael
>
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From matthew at corp.crocker.com  Mon Jun 20 18:25:01 2011
From: matthew at corp.crocker.com (Matthew S. Crocker)
Date: Mon, 20 Jun 2011 10:25:01 -0400 (EDT)
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <BANLkTikW0tjViHT6YZyb73B2GbXOCw3DOw@mail.gmail.com>
Message-ID: <09c0a579-6210-4db5-a669-5d9e2ca6beac@zimbra1.crocker.com>

I use Zhone MALCs for ADSL2+/POTS to the customer. It handles MGCP or SIP back to the softswitch. I'm using Broadsoft not Freeswitch but should still work just as well. 

http://www.zhone.com/products/MALC-XP/ 

-Matt 

----- Original Message -----

> From: "Nandy Dagondon" <gcd at i.ph>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Thursday, June 16, 2011 7:03:50 PM
> Subject: [Freeswitch-users] VoIP IP DSLAMs

> hello everybody,

> i'm looking for small (24/48 ports) IP DSLAMs that inter-connects
> directly to FreeSwitch via IP. is this already available on the
> market? or do we still have to connect FreeSwitch via POTS splitters
> and FXS gateways? i appreciate if you can mention some
> brands/models.

> tks,
> nandy

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

> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From matthew at corp.crocker.com  Mon Jun 20 18:26:58 2011
From: matthew at corp.crocker.com (Matthew S. Crocker)
Date: Mon, 20 Jun 2011 10:26:58 -0400 (EDT)
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <0013F8A2-8D82-470C-B8D8-45C22BDB3A00@ipeva.fr>
Message-ID: <84b3f6b0-9cff-40cc-8a32-f1a2da1458ab@zimbra1.crocker.com>

Zhone MALC-XP are nowhere near $11k. They are < $100/port 

----- Original Message -----

> From: "David Ponzone" <david.ponzone at ipeva.fr>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Friday, June 17, 2011 5:29:48 AM
> Subject: Re: [Freeswitch-users] VoIP IP DSLAMs

> Hmmm unlikely I think.
> The Zhone box for instance seems to be around 11,000$.

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

> Service Client IP eva

> tel: 0811 46 26 26

> www.ipeva.fr - www.ipeva-studio.com

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

> Le 17/06/2011 ? 10:25, Nandy Dagondon a ?crit :

> > yes, it doesn't add up because the application is in a rural area.
> > the Phybridge is a good solution except for the price. thanks
> > David.
> > however, the MSAN, perfectly describes what i'm looking - hopefully
> > the investment is lower.
> 

> > thanks to all who contributed.
> 

> > On Fri, Jun 17, 2011 at 3:37 PM, David Ponzone <
> > david.ponzone at ipeva.fr > wrote:
> 

> > > Nandy,
> > 
> 

> > > something does not add up.
> > 
> 
> > > When you deploy DSLAMs for public lines, it's not for 24/48
> > > ports,
> > > but rather for 20 000 to 30 000 ports.
> > 
> 
> > > Perhaps there is a confusion about what is a public line.
> > 
> 
> > > A public line is the 200 meters-8km pair of copper going from the
> > > local (legacy) telco facility to your house.
> > 
> 
> > > You don't install your own DSLAMs for 24/48 ports, or perhaps you
> > > are
> > > in a very specific situation.
> > 
> 
> > > I really doubt you will find a cost effective DSLAM for so few
> > > ports.
> > 
> 
> > > Plus, you realize you will need to install on DSLAM in all the
> > > telco
> > > facilities required to get the coverage you need
> > 
> 

> > > There are small DSLAMs on the market, but they are targeted for
> > > private lines (hospitality, health care, ...) and I really don't
> > > know if they can work on public lines.
> > 
> 

> > > In case you actually need a small DSLAM just to run ADSL over a
> > > private pair of copper, you should have a look at:
> > 
> 

> > > http://www.phybridge.com/uniphyer-ip-phones.aspx
> > 
> 

> > > This thing does not call itself a DSLAM but it is one.
> > 
> 
> > > They market that as a "switch" that enables LAN on your copper
> > > wires.
> > 
> 
> > > It's a small lie for data usage as you only get 25Mbps/1.4Mbps
> > > (ATM),
> > > but for voice, that's enough.
> > 
> 
> > > Basically, you put the "switch"-DSLAM in the network cabinet
> > > where
> > > the copper wires go, you connect the wires to it, you connect the
> > > ethernet port to your LAN (so your FS). In each room/office, you
> > > plug an adapter (provided also by Phybridge) which is basically a
> > > small ADSL modem (powered from the switch through the wire), from
> > > which you get an ethernet port, with POE on it!
> > 
> 

> > > I like the solution on the paper, but never used. Because of the
> > > cost:
> > 
> 
> > > a 24 ports switch is 3500$
> > 
> 
> > > one adapter is 122$
> > 
> 

> > > so a full 24 ports solution would be 6500$, which is not very
> > > competitive compared to recabling the place.
> > 
> 

> > > But your mileage may vary.
> > 
> 

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

> > > Service Client IP eva
> > 
> 

> > > tel: 0811 46 26 26
> > 
> 

> > > www.ipeva.fr - www.ipeva-studio.com
> > 
> 

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

> > > Le 17/06/2011 ? 08:28, Nandy Dagondon a ?crit :
> > 
> 

> > > > yes, without ATM layer and it's for public lines.
> > > 
> > 
> 

> > > > On Fri, Jun 17, 2011 at 2:19 PM, David Ponzone <
> > > > david.ponzone at ipeva.fr > wrote:
> > > 
> > 
> 

> > > > > Which density are you looking for ?
> > > > 
> > > 
> > 
> 
> > > > > Is it for use in a private building or on public lines ?
> > > > 
> > > 
> > 
> 

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

> > > > > Service Client IP eva
> > > > 
> > > 
> > 
> 

> > > > > tel: 0811 46 26 26
> > > > 
> > > 
> > 
> 

> > > > > www.ipeva.fr - www.ipeva-studio.com
> > > > 
> > > 
> > 
> 

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

> > > > > Le 17/06/2011 ? 08:07, Nandy Dagondon a ?crit :
> > > > 
> > > 
> > 
> 

> > > > > > hello guys!
> > > > > 
> > > > 
> > > 
> > 
> 

> > > > > > i'm trying to setup a small exchange w/ Internet service -
> > > > > > of
> > > > > > course,
> > > > > > using FS as the exchange. DSLAMs connect to POTS via 2-wire
> > > > > > splitters and that would require using FXS gateway (as what
> > > > > > Rob
> > > > > > mentioned).
> > > > > 
> > > > 
> > > 
> > 
> 

> > > > > > i'm trying to find out if there's a way we can omit the FXS
> > > > > > -
> > > > > > so
> > > > > > it's
> > > > > > FS-to-DSLAM direct via IP. i'm not sure if this setup is
> > > > > > now
> > > > > > possible on newer IP DSLAMs.
> > > > > 
> > > > 
> > > 
> > 
> 

> > > > > > On Fri, Jun 17, 2011 at 10:45 AM, jay binks <
> > > > > > jaybinks at gmail.com
> > > > > > >
> > > > > > wrote:
> > > > > 
> > > > 
> > > 
> > 
> 

> > > > > > > this is MY take on what Nandy is after.
> > > > > > 
> > > > > 
> > > > 
> > > 
> > 
> 

> > > > > > > A Dslam provides DSL on certain frequencies of the line..
> > > > > > 
> > > > > 
> > > > 
> > > 
> > 
> 
> > > > > > > ( some of ) the other frequencies are used for voice.
> > > > > > 
> > > > > 
> > > > 
> > > 
> > 
> 

> > > > > > > my understanding is that sometimes this is split off to
> > > > > > > another
> > > > > > > device to provide the voice,
> > > > > > 
> > > > > 
> > > > 
> > > 
> > 
> 
> > > > > > > but in this case Nandy is after a DSlam that will do the
> > > > > > > DSL
> > > > > > > part
> > > > > > > AND
> > > > > > > the voice frequencies ( Voice signalling / audio by SIP /
> > > > > > > RTP
> > > > > > > )
> > > > > > 
> > > > > 
> > > > 
> > > 
> > 
> 

> > > > > > > I guess this is a logical question and would be quite
> > > > > > > interesting
> > > > > > > to
> > > > > > > see if there are such devices..
> > > > > > 
> > > > > 
> > > > 
> > > 
> > 
> 
> > > > > > > sorry I dont know of any :(
> > > > > > 
> > > > > 
> > > > 
> > > 
> > 
> 

> > > > > > > Jay
> > > > > > 
> > > > > 
> > > > 
> > > 
> > 
> 

> > > > > > > On Fri, Jun 17, 2011 at 11:31 AM, Rob Hutton <
> > > > > > > justlikeef at gmail.com
> > > > > > > >
> > > > > > > wrote:
> > > > > > 
> > > > > 
> > > > 
> > > 
> > 
> 

> > > > > > > > Can you explain further what you are trying to do? A
> > > > > > > > DSLAM
> > > > > > > > is
> > > > > > > > a
> > > > > > > > device that provides DSL transport, which is
> > > > > > > > independent
> > > > > > > > of
> > > > > > > > what
> > > > > > > > you
> > > > > > > > run across it. So normally, you have
> > > > > > > 
> > > > > > 
> > > > > 
> > > > 
> > > 
> > 
> 

> > > > > > > > |Server | ----|network|----|DSLAM| --------[dsl line]
> > > > > > > > |-------[modem
> > > > > > > > |or modem/router combo]-----[network]---[Client]
> > > > > > > 
> > > > > > 
> > > > > 
> > > > 
> > > 
> > 
> 

> > > > > > > > What the Server is (Could be Freeswitch) and what the
> > > > > > > > Client
> > > > > > > > is
> > > > > > > > (could be a VOIP phone) are independent of the
> > > > > > > > transport???
> > > > > > > > Onewire,
> > > > > > > > Cisco, and others make some DSL Modem/Router combos
> > > > > > > > with
> > > > > > > > integrated
> > > > > > > > Voip to FXS ports for the CPE end.
> > > > > > > 
> > > > > > 
> > > > > 
> > > > 
> > > 
> > 
> 

> > > > > > > > If you find a DSLAM running one of the OSs that
> > > > > > > > Freeswitch
> > > > > > > > supports,
> > > > > > > > you could compile Freeswitch for it, but I haven't seen
> > > > > > > > one
> > > > > > > > with
> > > > > > > > enough CPU tp handle much besides the traffic.
> > > > > > > 
> > > > > > 
> > > > > 
> > > > 
> > > 
> > 
> 

> > > > > > > > On Thursday 16 June 2011 19:03:50 Nandy Dagondon wrote:
> > > > > > > 
> > > > > > 
> > > > > 
> > > > 
> > > 
> > 
> 
> > > > > > > > > hello everybody,
> > > > > > > 
> > > > > > 
> > > > > 
> > > > 
> > > 
> > 
> 
> > > > > > > > >
> > > > > > > 
> > > > > > 
> > > > > 
> > > > 
> > > 
> > 
> 
> > > > > > > > > i'm looking for small (24/48 ports) IP DSLAMs that
> > > > > > > > > inter-connects
> > > > > > > > > directly
> > > > > > > 
> > > > > > 
> > > > > 
> > > > 
> > > 
> > 
> 
> > > > > > > > > to FreeSwitch via IP. is this already available on
> > > > > > > > > the
> > > > > > > > > market?
> > > > > > > > > or
> > > > > > > > > do we
> > > > > > > 
> > > > > > 
> > > > > 
> > > > 
> > > 
> > 
> 
> > > > > > > > > still have to connect FreeSwitch via POTS splitters
> > > > > > > > > and
> > > > > > > > > FXS
> > > > > > > > > gateways? i
> > > > > > > 
> > > > > > 
> > > > > 
> > > > 
> > > 
> > 
> 
> > > > > > > > > appreciate if you can mention some brands/models.
> > > > > > > 
> > > > > > 
> > > > > 
> > > > 
> > > 
> > 
> 
> > > > > > > > >
> > > > > > > 
> > > > > > 
> > > > > 
> > > > 
> > > 
> > 
> 
> > > > > > > > > tks,
> > > > > > > 
> > > > > > 
> > > > > 
> > > > 
> > > 
> > 
> 
> > > > > > > > > nandy
> > > > > > > 
> > > > > > 
> > > > > 
> > > > 
> > > 
> > 
> 

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

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

> > > > > > > --
> > > > > > 
> > > > > 
> > > > 
> > > 
> > 
> 
> > > > > > > Sincerely
> > > > > > 
> > > > > 
> > > > 
> > > 
> > 
> 

> > > > > > > Jay
> > > > > > 
> > > > > 
> > > > 
> > > 
> > 
> 

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

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

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

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

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

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

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

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

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

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

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

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

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

> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From avi at avimarcus.net  Mon Jun 20 18:33:17 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 20 Jun 2011 17:33:17 +0300
Subject: [Freeswitch-users] Call Hangup
In-Reply-To: <4DFF12B8.5090008@seletech.com>
References: <4DFEFCE0.9040809@seletech.com>
	<B8D314BD-A38C-4B6D-AC95-84643B433E9C@ipeva.fr>
	<4DFF12B8.5090008@seletech.com>
Message-ID: <BANLkTinxDahbHqi5ACUev5cyj_b=sWR+tQ@mail.gmail.com>

You need to do a "sofia profile internal restart" to load your new settings.

 -Avi Marcus

On Mon, Jun 20, 2011 at 12:28 PM, Alessandro <a.luppi at seletech.com> wrote:

> **
> Hi,
>
> i set
> <param name="enable-timer" value="true"/>
> <param name="session-timeout" value="120"/>
>
> on sip profile, but doesn't change. FS doesn't hang up the call after one
> of the two extension disappear from the network.
> With wireshark i don't see re-invite during the call.
>
> Regards
>
> Alessandro
>   Il 20/06/2011 10:43, David Ponzone ha scritto:
>
> Alessandro,
>
>  there are 2 ways to do that:
> Session timers: with this, FS will send a RE-INVITE to phones every X
> minutes, and if the phone does not answer, it's considerer dead, so it is
> unregistered and the current calls will be hung up
> or
> RTP Timeout: this will detect that media is missing, and will hang up the
> call
>
>  Session timers has to be enabled on the SIP Profile with:
> <param name="enable-timer" value="true"/>
> <param name="session-timeout" value="120"/>
>
>  RTP timeout has to be enabled in the SIP Profile with:
>       <param name="rtp-timeout-sec" value="300"/>
>  (use the value you wish. Also, be careful with this, it can lead to
> issues with some phones when you put a call on hold/mute or if you use VAD)
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
>  Service Client IPeva
>  tel:      0811 46 26 26
>  www.ipeva.fr  -   www.ipeva-studio.com
>
>  *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
>  Le 20/06/2011 ? 09:55, Alessandro a ?crit :
>
>  Hi,
>
> I have an active call from the extension A to the extension B.
> If one of the two extension is disconnected from the network without
> deregistration the call isn't hung up by the server. The call should be
> hangup when FS realizes that the extension isn't in the network?
> Is there a time out to be set somewhere?
>
> thanks
>
> Regards
>
> Alessandro Luppi
>
> --
> Ing. Alessandro Luppi
> Software development
> Seletech srl
> Via Collodi 8, 20052 Monza (MI) - Italy
> Tel: +39.039.5962000 - Fax: +39.039.9716905
> email: a.luppi at seletech.com - Web: www.seletech.com   or   www.seletech.eu
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> Ing. Alessandro Luppi
> Software development
> Seletech srl
> Via Collodi 8, 20052 Monza (MI) - Italy
> Tel: +39.039.5962000 - Fax: +39.039.9716905
> email: a.luppi at seletech.com - Web: www.seletech.com   or   www.seletech.eu
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mario_fs at mgtech.com  Mon Jun 20 18:55:00 2011
From: mario_fs at mgtech.com (Mario G)
Date: Mon, 20 Jun 2011 07:55:00 -0700
Subject: [Freeswitch-users] FreeSWITCH Cookbook Recipe List
In-Reply-To: <BANLkTima7xQPi2JYe908CfyZaYmOMsw=aQ@mail.gmail.com>
References: <BANLkTima7xQPi2JYe908CfyZaYmOMsw=aQ@mail.gmail.com>
Message-ID: <7EAFE942-322C-4EC0-941A-F1226B8DDAD1@mgtech.com>

I see there is a need for install on OS X Server. I thought the OS X install I did for the wiki was quite comprehensive. What is missing? It should be the same for OS X server. Also, I will do a clean install on OS X 10.7 next month to update the wiki.

On Jun 17, 2011, at 8:14 PM, Michael Collins wrote:

> Hello all!
> 
> Some have inquired recently about what's needed to complete the FS cookbook. I've uploaded the most recent list of recipes that we have identified. Some are already written but many are yet to be done. Check out this page:
> 
> http://wiki.freeswitch.org/wiki/Cookbook#Information_for_prospective_authors
> 
> Contact me if you have any questions about helping with any of these recipes.
> 
> Thanks,
> Michael
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From skyptester at gmail.com  Sun Jun 19 15:34:17 2011
From: skyptester at gmail.com (Skyp Tester)
Date: Sun, 19 Jun 2011 14:34:17 +0300
Subject: [Freeswitch-users] Skypopen,
	Skype 2.2.0.35 Beta for Linux and Skype Connect
In-Reply-To: <BANLkTim+WjTGiTFNXBdTLqSZ_0t-DVhhmA@mail.gmail.com>
References: <BANLkTim+WjTGiTFNXBdTLqSZ_0t-DVhhmA@mail.gmail.com>
Message-ID: <BANLkTi=ru21gP4nLpSKSLqMzwBqzwaWHHQ@mail.gmail.com>

Hi



There is problem with Skype Linux client 2.0.0.72 , I cant call to Skype
Connect client, error ?FAILUREREASON 2? = ?User or phone number does not
exist?.

http://www.skype.com/intl/en-us/business/skype-connect/



Installed skypopen with Skype client 2.0.0.72 .

Upgraded to 2.2.0.35:

1)      install libstdc++-4.3.2-7.i386.rpm

2)      copy over /usr/bin/skype and /usr/share/skype/lang with new version



Now I can call to Skype Connect client but there are some problems

1)      audio is one way, Skype->FS

2)      after first call there is error ?NO SPACE WRITE: 640? and no more
Skype calls, ?ERROR 68? & ?If I don't connect immediately, please give the
Skype client authorization to be connected by Skypopen  (and to not ask you
again)?



Can we hope support to Skype Linux 2.2.0.35 or there is some big technical
obstacle?





Centos 5.6 x86_64

2.6.18-238.12.1.el5 #1 SMP

FreeSWITCH Version 1.0.head (git-fccbba5 2011-05-18 19-00-42 -0400)
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EXECUTE sofia/internal/1001 at fsserver bridge(skypopen/skype101/echo123)
2011-06-19 12:53:47.517153 [DEBUG] mod_skypopen.c:1264         [ac7ca63|9350fb9] [DEBUG_SKYPE  1264 ][none           ][N/A,N/A] 1 SESSION_REQUEST 3cc797ba-7552-459c-9a86-9daa8745821c
2011-06-19 12:53:47.517153 [DEBUG] mod_skypopen.c:1299         [ac7ca63|9350fb9] [DEBUG_SKYPE  1299 ][none           ][N/A,N/A] globals.SKYPOPEN_INTERFACES[1].name=|||skype101|||?
2011-06-19 12:53:47.517153 [DEBUG] mod_skypopen.c:244          [ac7ca63|9350fb9] [DEBUG_SKYPE  244  ][skype101       ][IDLE,IDLE] codecs UP
2011-06-19 12:53:47.517153 [DEBUG] mod_skypopen.c:273          [ac7ca63|9350fb9] [DEBUG_SKYPE  273  ][skype101       ][IDLE,IDLE] skypopen_tech_init SUCCESS
2011-06-19 12:53:47.517153 [NOTICE] switch_channel.c:816 New Channel skypopen/skype101/echo123 [3cc797ba-7552-459c-9a86-9daa8745821c]
2011-06-19 12:53:47.517153 [DEBUG] mod_skypopen.c:1371 (skypopen/skype101/echo123) State Change CS_NEW -> CS_INIT
2011-06-19 12:53:47.517153 [DEBUG] switch_core_session.c:1114 Send signal skypopen/skype101/echo123 [BREAK]
2011-06-19 12:53:47.517153 [DEBUG] mod_skypopen.c:728          [ac7ca63|9350fb9] [DEBUG_SKYPE  728  ][skype101       ][IDLE,IDLE] skypopen/skype101/echo123 CHANNEL got SWITCH_SIG_BREAK
2011-06-19 12:53:47.517153 [DEBUG] skypopen_protocol.c:1143    [ac7ca63|9350fb9] [DEBUG_SKYPE  1143 ][skype101       ][IDLE,IDLE] Calling Skype, rdest is: echo123
2011-06-19 12:53:47.517153 [DEBUG] skypopen_protocol.c:1619    [ac7ca63|9350fb9] [DEBUG_SKYPE  1619 ][skype101       ][IDLE,IDLE] SENDING: |||CALL echo123||||
2011-06-19 12:53:47.517153 [DEBUG] switch_core_state_machine.c:325 (skypopen/skype101/echo123) Running State Change CS_INIT
2011-06-19 12:53:47.517153 [DEBUG] switch_core_state_machine.c:361 (skypopen/skype101/echo123) State INIT
2011-06-19 12:53:47.517153 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][IDLE,IDLE] READING: |||CALL 112 STATUS UNPLACED||| 
2011-06-19 12:53:47.517153 [DEBUG] skypopen_protocol.c:714     [ac7ca63|9350fb9] [DEBUG_SKYPE  714  ][skype101       ][DIALING,UNPLACD] skype_call: 112 is now UNPLACED
2011-06-19 12:53:47.517153 [DEBUG] mod_skypopen.c:463 (skypopen/skype101/echo123) State Change CS_INIT -> CS_ROUTING
2011-06-19 12:53:47.517153 [DEBUG] switch_core_session.c:1114 Send signal skypopen/skype101/echo123 [BREAK]
2011-06-19 12:53:47.517153 [DEBUG] mod_skypopen.c:728          [ac7ca63|9350fb9] [DEBUG_SKYPE  728  ][skype101       ][DIALING,UNPLACD] skypopen/skype101/echo123 CHANNEL got SWITCH_SIG_BREAK
2011-06-19 12:53:47.517153 [DEBUG] mod_skypopen.c:464          [ac7ca63|9350fb9] [DEBUG_SKYPE  464  ][skype101       ][DIALING,UNPLACD] skype101 CHANNEL INIT 3cc797ba-7552-459c-9a86-9daa8745821c
2011-06-19 12:53:47.517153 [DEBUG] switch_core_state_machine.c:361 (skypopen/skype101/echo123) State INIT going to sleep
2011-06-19 12:53:47.517153 [DEBUG] switch_core_state_machine.c:325 (skypopen/skype101/echo123) Running State Change CS_ROUTING
2011-06-19 12:53:47.517153 [DEBUG] switch_channel.c:1687 (skypopen/skype101/echo123) Callstate Change DOWN -> RINGING
2011-06-19 12:53:47.517153 [DEBUG] switch_core_state_machine.c:364 (skypopen/skype101/echo123) State ROUTING
2011-06-19 12:53:47.517153 [DEBUG] mod_skypopen.c:665          [ac7ca63|9350fb9] [DEBUG_SKYPE  665  ][skype101       ][DIALING,UNPLACD] skype101 CHANNEL ROUTING
2011-06-19 12:53:47.517153 [DEBUG] switch_ivr_originate.c:66 (skypopen/skype101/echo123) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2011-06-19 12:53:47.517153 [DEBUG] switch_core_session.c:1114 Send signal skypopen/skype101/echo123 [BREAK]
2011-06-19 12:53:47.517153 [DEBUG] mod_skypopen.c:728          [ac7ca63|9350fb9] [DEBUG_SKYPE  728  ][skype101       ][DIALING,UNPLACD] skypopen/skype101/echo123 CHANNEL got SWITCH_SIG_BREAK
2011-06-19 12:53:47.517153 [DEBUG] switch_core_state_machine.c:364 (skypopen/skype101/echo123) State ROUTING going to sleep
2011-06-19 12:53:47.517153 [DEBUG] switch_core_state_machine.c:325 (skypopen/skype101/echo123) Running State Change CS_CONSUME_MEDIA
2011-06-19 12:53:47.517153 [DEBUG] switch_core_state_machine.c:383 (skypopen/skype101/echo123) State CONSUME_MEDIA
2011-06-19 12:53:47.517153 [DEBUG] mod_skypopen.c:748          [ac7ca63|9350fb9] [DEBUG_SKYPE  748  ][skype101       ][DIALING,UNPLACD] skype101 CHANNEL CONSUME_MEDIA
2011-06-19 12:53:47.517153 [DEBUG] switch_core_state_machine.c:383 (skypopen/skype101/echo123) State CONSUME_MEDIA going to sleep
2011-06-19 12:53:47.565225 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][DIALING,UNPLACD] READING: |||CALL 112 STATUS ROUTING||| 
2011-06-19 12:53:47.565225 [DEBUG] skypopen_protocol.c:709     [ac7ca63|9350fb9] [DEBUG_SKYPE  709  ][skype101       ][DIALING,ROUTING] skype_call: 112 is now ROUTING
2011-06-19 12:53:47.565225 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][DIALING,ROUTING] READING: |||CALL 112 STATUS ROUTING||| 
2011-06-19 12:53:47.565225 [DEBUG] skypopen_protocol.c:709     [ac7ca63|9350fb9] [DEBUG_SKYPE  709  ][skype101       ][DIALING,ROUTING] skype_call: 112 is now ROUTING
2011-06-19 12:53:47.626189 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][DIALING,ROUTING] READING: |||CALL 112 STATUS RINGING||| 
2011-06-19 12:53:47.626189 [DEBUG] skypopen_protocol.c:612     [ac7ca63|9350fb9] [DEBUG_SKYPE  612  ][skype101       ][RINGING,RINGING] Our remote party in skype_call 112 is RINGING
2011-06-19 12:53:47.626189 [NOTICE] mod_skypopen.c:2309 Ring-Ready skypopen/skype101/echo123!
2011-06-19 12:53:47.626189 [DEBUG] mod_skypopen.c:2310         [ac7ca63|9350fb9] [DEBUG_SKYPE  2310 ][skype101       ][RINGING,RINGING] skype_call: REMOTE PARTY RINGING
2011-06-19 12:53:49.326423 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][RINGING,RINGING] READING: |||CALL 112 STATUS INPROGRESS||| 
2011-06-19 12:53:49.326423 [DEBUG] skypopen_protocol.c:725     [ac7ca63|9350fb9] [DEBUG_SKYPE  725  ][skype101       ][RINGING,RINGING] skype_call: 112 is now active
2011-06-19 12:53:49.326423 [DEBUG] skypopen_protocol.c:732     [ac7ca63|9350fb9] [DEBUG_SKYPE  732  ][skype101       ][UP,INPROGRS] START start_audio_threads
2011-06-19 12:53:49.326423 [DEBUG] mod_skypopen.c:2208         [ac7ca63|9350fb9] [DEBUG_SKYPE  2208 ][skype101       ][UP,INPROGRS] started tcp_srv_thread thread.
2011-06-19 12:53:49.326423 [DEBUG] mod_skypopen.c:2225         [ac7ca63|9350fb9] [DEBUG_SKYPE  2225 ][skype101       ][UP,INPROGRS] started tcp_cli_thread thread.
2011-06-19 12:53:49.326423 [DEBUG] skypopen_protocol.c:89      [ac7ca63|9350fb9] [DEBUG_SKYPE  89   ][skype101       ][UP,INPROGRS] Binded! *which_port=32775, tech_pvt->tcp_cli_port=32774, tech_pvt->tcp_srv_port=32775
2011-06-19 12:53:49.326423 [DEBUG] skypopen_protocol.c:94      [ac7ca63|9350fb9] [DEBUG_SKYPE  94   ][skype101       ][UP,INPROGRS] 1 SO_RCVBUF is 87380, size is 4
2011-06-19 12:53:49.326423 [DEBUG] skypopen_protocol.c:98      [ac7ca63|9350fb9] [DEBUG_SKYPE  98   ][skype101       ][UP,INPROGRS] 1 SO_SNDBUF is 16384, size is 4
2011-06-19 12:53:49.326423 [DEBUG] skypopen_protocol.c:115     [ac7ca63|9350fb9] [DEBUG_SKYPE  115  ][skype101       ][UP,INPROGRS] 2 SO_RCVBUF is 5120, size is 4
2011-06-19 12:53:49.326423 [DEBUG] skypopen_protocol.c:130     [ac7ca63|9350fb9] [DEBUG_SKYPE  130  ][skype101       ][UP,INPROGRS] 2 SO_SNDBUF is 5120, size is 4
2011-06-19 12:53:49.326423 [DEBUG] skypopen_protocol.c:134     [ac7ca63|9350fb9] [DEBUG_SKYPE  134  ][skype101       ][UP,INPROGRS] TCP_NODELAY is 0
2011-06-19 12:53:49.326423 [DEBUG] skypopen_protocol.c:141     [ac7ca63|9350fb9] [DEBUG_SKYPE  141  ][skype101       ][UP,INPROGRS] TCP_NODELAY is 1
2011-06-19 12:53:49.326423 [DEBUG] skypopen_protocol.c:833     [ac7ca63|9350fb9] [DEBUG_SKYPE  833  ][skype101       ][UP,INPROGRS] started tcp_srv_thread thread.
2011-06-19 12:53:49.326423 [DEBUG] skypopen_protocol.c:89      [ac7ca63|9350fb9] [DEBUG_SKYPE  89   ][skype101       ][UP,INPROGRS] Binded! *which_port=32776, tech_pvt->tcp_cli_port=32776, tech_pvt->tcp_srv_port=32775
2011-06-19 12:53:49.326423 [DEBUG] skypopen_protocol.c:94      [ac7ca63|9350fb9] [DEBUG_SKYPE  94   ][skype101       ][UP,INPROGRS] 1 SO_RCVBUF is 87380, size is 4
2011-06-19 12:53:49.326423 [DEBUG] skypopen_protocol.c:98      [ac7ca63|9350fb9] [DEBUG_SKYPE  98   ][skype101       ][UP,INPROGRS] 1 SO_SNDBUF is 16384, size is 4
2011-06-19 12:53:49.326423 [DEBUG] skypopen_protocol.c:115     [ac7ca63|9350fb9] [DEBUG_SKYPE  115  ][skype101       ][UP,INPROGRS] 2 SO_RCVBUF is 5120, size is 4
2011-06-19 12:53:49.326423 [DEBUG] skypopen_protocol.c:130     [ac7ca63|9350fb9] [DEBUG_SKYPE  130  ][skype101       ][UP,INPROGRS] 2 SO_SNDBUF is 5120, size is 4
2011-06-19 12:53:49.326423 [DEBUG] skypopen_protocol.c:134     [ac7ca63|9350fb9] [DEBUG_SKYPE  134  ][skype101       ][UP,INPROGRS] TCP_NODELAY is 0
2011-06-19 12:53:49.326423 [DEBUG] skypopen_protocol.c:141     [ac7ca63|9350fb9] [DEBUG_SKYPE  141  ][skype101       ][UP,INPROGRS] TCP_NODELAY is 1
2011-06-19 12:53:49.326423 [DEBUG] skypopen_protocol.c:997     [ac7ca63|9350fb9] [DEBUG_SKYPE  997  ][skype101       ][UP,INPROGRS] started tcp_cli_thread thread.
2011-06-19 12:53:49.426437 [DEBUG] skypopen_protocol.c:1619    [ac7ca63|9350fb9] [DEBUG_SKYPE  1619 ][skype101       ][UP,INPROGRS] SENDING: |||ALTER CALL 112 SET_INPUT PORT="32776"||||
2011-06-19 12:53:49.426437 [DEBUG] skypopen_protocol.c:1619    [ac7ca63|9350fb9] [DEBUG_SKYPE  1619 ][skype101       ][UP,INPROGRS] SENDING: |||#output ALTER CALL 112 SET_OUTPUT PORT="32775"||||
2011-06-19 12:53:49.426437 [DEBUG] skypopen_protocol.c:2019    [ac7ca63|9350fb9] [DEBUG_SKYPE  2019 ][skype101       ][UP,INPROGRS] Outbound Channel Answered! session_uuid_str=3cc797ba-7552-459c-9a86-9daa8745821c
2011-06-19 12:53:49.426437 [DEBUG] switch_channel.c:2859 (skypopen/skype101/echo123) Callstate Change RINGING -> ACTIVE
2011-06-19 12:53:49.426437 [NOTICE] mod_skypopen.c:2369 Channel [skypopen/skype101/echo123] has been answered
2011-06-19 12:53:49.426437 [DEBUG] mod_skypopen.c:1149         [ac7ca63|9350fb9] [DEBUG_SKYPE  1149 ][skype101       ][UP,INPROGRS] skypopen/skype101/echo123 CHANNEL got SWITCH_MESSAGE_INDICATE_AUDIO_SYNC
2011-06-19 12:53:49.426437 [DEBUG] mod_skypopen.c:1171         [ac7ca63|9350fb9] [DEBUG_SKYPE  1171 ][skype101       ][UP,INPROGRS] Synching audio
2011-06-19 12:53:49.426437 [DEBUG] mod_skypopen.c:2377         [ac7ca63|9350fb9] [DEBUG_SKYPE  2377 ][skype101       ][UP,INPROGRS] outbound_channel_answered!
2011-06-19 12:53:49.426437 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 0||| 
2011-06-19 12:53:49.426437 [DEBUG] skypopen_protocol.c:564     [ac7ca63|9350fb9] [DEBUG_SKYPE  564  ][skype101       ][UP,INPROGRS] Synching audio on skype_call: 112.
2011-06-19 12:53:49.426437 [DEBUG] skypopen_protocol.c:862     [ac7ca63|9350fb9] [DEBUG_SKYPE  862  ][skype101       ][UP,INPROGRS] ACCEPTED here I send you 32775
2011-06-19 12:53:49.426437 [DEBUG] skypopen_protocol.c:867     [ac7ca63|9350fb9] [DEBUG_SKYPE  867  ][skype101       ][UP,INPROGRS] 3 SO_RCVBUF is 5120, size is 4
2011-06-19 12:53:49.426437 [DEBUG] skypopen_protocol.c:871     [ac7ca63|9350fb9] [DEBUG_SKYPE  871  ][skype101       ][UP,INPROGRS] 3 SO_SNDBUF is 5120, size is 4
2011-06-19 12:53:49.426437 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||ALTER CALL 112 SET_INPUT PORT="32776"||| 
2011-06-19 12:53:49.426437 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||#output ALTER CALL 112 SET_OUTPUT PORT="32775"||| 
2011-06-19 12:53:49.426437 [DEBUG] skypopen_protocol.c:1026    [ac7ca63|9350fb9] [DEBUG_SKYPE  1026 ][skype101       ][UP,INPROGRS] ACCEPTED here you send me 32776
2011-06-19 12:53:49.426437 [DEBUG] skypopen_protocol.c:1031    [ac7ca63|9350fb9] [DEBUG_SKYPE  1031 ][skype101       ][UP,INPROGRS] 4 SO_RCVBUF is 5120, size is 4
2011-06-19 12:53:49.426437 [DEBUG] skypopen_protocol.c:1035    [ac7ca63|9350fb9] [DEBUG_SKYPE  1035 ][skype101       ][UP,INPROGRS] 4 SO_SNDBUF is 5120, size is 4
2011-06-19 12:53:49.426437 [DEBUG] switch_ivr_originate.c:3206 Originate Resulted in Success: [skypopen/skype101/echo123]
2011-06-19 12:53:49.426437 [DEBUG] mod_skypopen.c:1149         [ac7ca63|9350fb9] [DEBUG_SKYPE  1149 ][skype101       ][UP,INPROGRS] skypopen/skype101/echo123 CHANNEL got SWITCH_MESSAGE_INDICATE_AUDIO_SYNC
2011-06-19 12:53:49.426437 [DEBUG] mod_skypopen.c:1171         [ac7ca63|9350fb9] [DEBUG_SKYPE  1171 ][skype101       ][UP,INPROGRS] Synching audio
2011-06-19 12:53:49.426437 [DEBUG] mod_skypopen.c:1174         [ac7ca63|9350fb9] [DEBUG_SKYPE  1174 ][skype101       ][UP,INPROGRS] skypopen/skype101/echo123 CHANNEL got SWITCH_MESSAGE_INDICATE_BRIDGE
2011-06-19 12:53:49.426437 [DEBUG] mod_skypopen.c:1196         [ac7ca63|9350fb9] [DEBUG_SKYPE  1196 ][skype101       ][UP,INPROGRS] Synching audio
2011-06-19 12:53:49.426437 [DEBUG] switch_core_session.c:707 Send signal skypopen/skype101/echo123 [BREAK]
2011-06-19 12:53:49.426437 [DEBUG] mod_skypopen.c:728          [ac7ca63|9350fb9] [DEBUG_SKYPE  728  ][skype101       ][UP,INPROGRS] skypopen/skype101/echo123 CHANNEL got SWITCH_SIG_BREAK
2011-06-19 12:53:49.426437 [DEBUG] switch_ivr_bridge.c:1239 (skypopen/skype101/echo123) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2011-06-19 12:53:49.426437 [DEBUG] switch_core_session.c:1114 Send signal skypopen/skype101/echo123 [BREAK]
2011-06-19 12:53:49.426437 [DEBUG] mod_skypopen.c:728          [ac7ca63|9350fb9] [DEBUG_SKYPE  728  ][skype101       ][UP,INPROGRS] skypopen/skype101/echo123 CHANNEL got SWITCH_SIG_BREAK
2011-06-19 12:53:49.426437 [DEBUG] switch_core_state_machine.c:325 (skypopen/skype101/echo123) Running State Change CS_EXCHANGE_MEDIA
2011-06-19 12:53:49.446447 [DEBUG] switch_core_state_machine.c:374 (skypopen/skype101/echo123) State EXCHANGE_MEDIA
2011-06-19 12:53:49.446447 [DEBUG] mod_skypopen.c:757          [ac7ca63|9350fb9] [DEBUG_SKYPE  757  ][skype101       ][UP,INPROGRS] skype101 CHANNEL EXCHANGE_MEDIA
2011-06-19 12:53:49.466445 [DEBUG] mod_skypopen.c:899          [ac7ca63|9350fb9] [DEBUG_SKYPE  899  ][skype101       ][UP,INPROGRS] CHANNEL READ FRAME goto CNG
2011-06-19 12:53:49.546455 [DEBUG] mod_skypopen.c:1202         [ac7ca63|9350fb9] [DEBUG_SKYPE  1202 ][skype101       ][UP,INPROGRS] MSG_ID=19
2011-06-19 12:53:49.606499 [DEBUG] switch_core_session.c:769 Send signal skypopen/skype101/echo123 [BREAK]
2011-06-19 12:53:49.606499 [DEBUG] mod_skypopen.c:728          [ac7ca63|9350fb9] [DEBUG_SKYPE  728  ][skype101       ][UP,INPROGRS] skypopen/skype101/echo123 CHANNEL got SWITCH_SIG_BREAK
2011-06-19 12:53:49.626465 [DEBUG] mod_skypopen.c:899          [ac7ca63|9350fb9] [DEBUG_SKYPE  899  ][skype101       ][UP,INPROGRS] CHANNEL READ FRAME goto CNG
2011-06-19 12:53:49.626465 [DEBUG] mod_skypopen.c:1202         [ac7ca63|9350fb9] [DEBUG_SKYPE  1202 ][skype101       ][UP,INPROGRS] MSG_ID=18
2011-06-19 12:53:50.326567 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 1||| 
2011-06-19 12:53:51.332778 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 2||| 
2011-06-19 12:53:52.328865 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 3||| 
2011-06-19 12:53:53.329050 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 4||| 
2011-06-19 12:53:54.230149 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 5||| 
2011-06-19 12:53:55.330298 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 6||| 
2011-06-19 12:53:56.330469 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 7||| 
2011-06-19 12:53:57.330619 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 8||| 
2011-06-19 12:53:58.333749 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 9||| 
2011-06-19 12:53:59.333919 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 10||| 
2011-06-19 12:54:00.334051 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 11||| 
2011-06-19 12:54:01.334213 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 12||| 
2011-06-19 12:54:02.334356 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 13||| 
2011-06-19 12:54:03.336467 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 14||| 
2011-06-19 12:54:04.323615 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 15||| 
2011-06-19 12:54:05.237748 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 16||| 
2011-06-19 12:54:06.317892 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 17||| 
2011-06-19 12:54:07.219050 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 18||| 
2011-06-19 12:54:08.319186 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 19||| 
2011-06-19 12:54:09.219400 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 20||| 
2011-06-19 12:54:09.219400 [DEBUG] skypopen_protocol.c:564     [ac7ca63|9350fb9] [DEBUG_SKYPE  564  ][skype101       ][UP,INPROGRS] Synching audio on skype_call: 112.
2011-06-19 12:54:10.319478 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 21||| 
2011-06-19 12:54:11.323629 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 22||| 
2011-06-19 12:54:12.324783 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 23||| 
2011-06-19 12:54:13.324950 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 24||| 
2011-06-19 12:54:14.225078 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 25||| 
2011-06-19 12:54:15.325218 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 26||| 
2011-06-19 12:54:16.325388 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 27||| 
2011-06-19 12:54:17.234497 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 28||| 
2011-06-19 12:54:18.329677 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 29||| 
2011-06-19 12:54:19.329836 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 30||| 
2011-06-19 12:54:20.229955 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 31||| 
2011-06-19 12:54:21.230081 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 32||| 
2011-06-19 12:54:22.230227 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 33||| 
2011-06-19 12:54:23.230396 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 34||| 
2011-06-19 12:54:24.317565 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][UP,INPROGRS] READING: |||CALL 112 DURATION 35||| 
2011-06-19 12:54:31.547588 [DEBUG] skypopen_protocol.c:1093    [ac7ca63|9350fb9] [DEBUG_SKYPE  1093 ][skype101       ][UP,INPROGRS] len=-1, error: Connection reset by peer
2011-06-19 12:54:31.547588 [DEBUG] skypopen_protocol.c:1103    [ac7ca63|9350fb9] [DEBUG_SKYPE  1103 ][skype101       ][UP,INPROGRS] Skype outbound audio GONE
2011-06-19 12:54:31.547588 [DEBUG] skypopen_protocol.c:1112    [ac7ca63|9350fb9] [DEBUG_SKYPE  1112 ][skype101       ][UP,INC_HNG] outbound audio server (I am it) EXITING
2011-06-19 12:54:31.567590 [DEBUG] skypopen_protocol.c:951     [ac7ca63|9350fb9] [DEBUG_SKYPE  951  ][skype101       ][UP,INC_HNG] Skype incoming audio GONE
2011-06-19 12:54:31.567590 [DEBUG] skypopen_protocol.c:960     [ac7ca63|9350fb9] [DEBUG_SKYPE  960  ][skype101       ][UP,INC_HNG] incoming audio (read) server (I am it) EXITING
2011-06-19 12:54:31.607588 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:31.667597 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:31.727606 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:31.787614 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:31.847657 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:31.907632 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:31.967640 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:32.027650 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:32.087658 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:32.147697 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:32.207676 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:32.267696 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:32.327694 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:32.387703 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:32.447724 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:32.507728 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:32.567737 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:32.627745 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:32.687755 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:32.747803 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:32.807772 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:32.867781 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:32.927790 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:32.987798 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:33.047851 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:33.107816 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:33.167825 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:33.227835 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:33.287842 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:33.347891 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:33.407860 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:33.467868 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:33.527878 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:33.587886 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:33.647936 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:33.707903 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:33.767995 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:33.827921 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:33.887930 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:33.947985 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:34.007947 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:34.067956 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:34.127965 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:34.187973 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:34.248014 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:34.307991 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:34.368073 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:34.428009 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:34.488018 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:34.548057 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:34.608035 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:34.668044 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:34.728052 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:34.788062 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:34.848105 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:34.908078 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:34.968088 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:35.028097 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:35.088105 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:35.148144 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:35.208121 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:35.268131 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:35.328140 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:35.388148 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:35.448195 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:35.508167 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:35.568175 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:35.628183 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:35.688192 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:35.748231 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:35.808210 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:35.868219 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:35.928228 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:35.988236 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:36.048277 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:36.109254 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:36.169263 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:36.229271 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:36.289281 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:36.350300 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:36.411298 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:36.472307 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:36.554320 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:36.615328 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:36.676338 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:36.737346 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:36.798356 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:36.859364 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:36.920373 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:36.981382 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:37.044390 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:37.106406 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:37.166408 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:37.226420 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:37.286427 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:37.346435 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:37.406444 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:37.466452 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:37.526465 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:37.586469 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:37.646479 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:37.706487 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:37.766497 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:37.826508 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:37.886514 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:37.946523 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:38.006531 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:38.066541 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:38.126606 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
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2011-06-19 12:54:38.426652 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:38.486601 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:38.546610 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:38.606619 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:38.666628 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:38.726641 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:38.786645 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:38.846654 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:38.906662 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:38.966672 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:39.026686 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:39.086689 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:39.146697 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:39.206707 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:39.266715 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
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2011-06-19 12:54:39.386732 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
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2011-06-19 12:54:39.506751 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:39.566759 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:39.626772 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:39.686777 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:39.746785 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:39.806794 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:39.866804 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:39.926837 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:39.986821 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:40.046829 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:40.106838 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:40.166847 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:40.226859 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:40.286864 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:40.346873 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:40.406882 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:40.466918 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:40.526903 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:40.586908 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:40.646917 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:40.706926 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:40.766935 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:40.826949 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:40.886952 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:40.946962 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:41.006970 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:41.066978 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:41.126991 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:41.186995 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:41.247004 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:41.307014 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
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2011-06-19 12:54:44.309442 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:44.369674 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:44.429459 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:44.489471 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:44.549478 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:44.609486 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:44.669495 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:44.729504 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:44.789512 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:44.849521 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:44.909530 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:44.969539 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:45.029548 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:45.089556 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:45.149566 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:45.209576 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:45.269582 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:45.329591 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:45.389600 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:45.449610 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:45.509619 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:45.569627 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:45.629635 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:45.689645 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:45.749655 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:45.809662 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:45.869671 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:45.929680 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:45.989687 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:46.049698 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:46.109707 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:46.169715 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:46.229724 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:46.289733 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:46.349742 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:46.409750 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:46.469758 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:46.529768 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:46.589778 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:46.649785 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:46.709794 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:46.769803 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:46.829811 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:46.889820 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:46.949828 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:47.009837 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:47.069845 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:47.129855 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:47.189863 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:47.249872 [DEBUG] mod_skypopen.c:1018         [ac7ca63|9350fb9] [DEBUG_SKYPE  1018 ][skype101       ][UP,INC_HNG] NO SPACE WRITE: 640
2011-06-19 12:54:47.329895 [DEBUG] switch_ivr_bridge.c:604 Send signal skypopen/skype101/echo123 [BREAK]
2011-06-19 12:54:47.329895 [DEBUG] mod_skypopen.c:728          [ac7ca63|9350fb9] [DEBUG_SKYPE  728  ][skype101       ][UP,INC_HNG] skypopen/skype101/echo123 CHANNEL got SWITCH_SIG_BREAK
2011-06-19 12:54:47.329895 [DEBUG] mod_skypopen.c:899          [ac7ca63|9350fb9] [DEBUG_SKYPE  899  ][skype101       ][UP,INC_HNG] CHANNEL READ FRAME goto CNG
2011-06-19 12:54:47.329895 [DEBUG] switch_ivr_bridge.c:584 BRIDGE THREAD DONE [skypopen/skype101/echo123]
2011-06-19 12:54:47.329895 [DEBUG] switch_channel.c:2592 (skypopen/skype101/echo123) Callstate Change ACTIVE -> HANGUP
2011-06-19 12:54:47.329895 [NOTICE] switch_ivr_bridge.c:656 Hangup skypopen/skype101/echo123 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2011-06-19 12:54:47.329895 [DEBUG] switch_channel.c:2608 Send signal skypopen/skype101/echo123 [KILL]
2011-06-19 12:54:47.329895 [DEBUG] mod_skypopen.c:701          [ac7ca63|9350fb9] [DEBUG_SKYPE  701  ][skype101       ][UP,INC_HNG] skypopen/skype101/echo123 CHANNEL got SWITCH_SIG_KILL
2011-06-19 12:54:47.329895 [DEBUG] switch_core_session.c:1114 Send signal skypopen/skype101/echo123 [BREAK]
2011-06-19 12:54:47.329895 [DEBUG] mod_skypopen.c:728          [ac7ca63|9350fb9] [DEBUG_SKYPE  728  ][skype101       ][HANG_RQ,INC_HNG] skypopen/skype101/echo123 CHANNEL got SWITCH_SIG_BREAK
2011-06-19 12:54:47.329895 [DEBUG] switch_core_state_machine.c:374 (skypopen/skype101/echo123) State EXCHANGE_MEDIA going to sleep
2011-06-19 12:54:47.329895 [DEBUG] switch_core_state_machine.c:325 (skypopen/skype101/echo123) Running State Change CS_HANGUP
2011-06-19 12:54:47.329895 [DEBUG] switch_core_state_machine.c:565 (skypopen/skype101/echo123) State HANGUP
2011-06-19 12:54:47.329895 [DEBUG] mod_skypopen.c:624          [ac7ca63|9350fb9] [DEBUG_SKYPE  624  ][skype101       ][HANG_RQ,INC_HNG] hanging up skype call: 112
2011-06-19 12:54:47.329895 [DEBUG] skypopen_protocol.c:1619    [ac7ca63|9350fb9] [DEBUG_SKYPE  1619 ][skype101       ][HANG_RQ,INC_HNG] SENDING: |||ALTER CALL 112 END HANGUP||||
2011-06-19 12:54:47.329895 [DEBUG] skypopen_protocol.c:1619    [ac7ca63|9350fb9] [DEBUG_SKYPE  1619 ][skype101       ][HANG_RQ,INC_HNG] SENDING: |||ALTER CALL 112 HANGUP||||
2011-06-19 12:54:47.329895 [DEBUG] mod_skypopen.c:630          [ac7ca63|9350fb9] [DEBUG_SKYPE  630  ][skype101       ][HANG_RQ,INC_HNG] skype101 CHANNEL HANGUP
2011-06-19 12:54:47.329895 [DEBUG] mod_skypopen.c:649 (skypopen/skype101/echo123) State Change CS_HANGUP -> CS_DESTROY
2011-06-19 12:54:47.329895 [DEBUG] switch_core_session.c:1114 Send signal skypopen/skype101/echo123 [BREAK]
2011-06-19 12:54:47.329895 [DEBUG] mod_skypopen.c:728          [ac7ca63|9350fb9] [DEBUG_SKYPE  728  ][skype101       ][DOWN,INC_HNG] skypopen/skype101/echo123 CHANNEL got SWITCH_SIG_BREAK
2011-06-19 12:54:47.329895 [DEBUG] switch_core_state_machine.c:565 (skypopen/skype101/echo123) State HANGUP going to sleep
2011-06-19 12:54:47.329895 [DEBUG] switch_core_session.c:1286 Session 8 (skypopen/skype101/echo123) Locked, Waiting on external entities
2011-06-19 12:54:47.329895 [NOTICE] switch_core_session.c:1304 Session 8 (skypopen/skype101/echo123) Ended
2011-06-19 12:54:47.329895 [NOTICE] switch_core_session.c:1306 Close Channel skypopen/skype101/echo123 [CS_DESTROY]
2011-06-19 12:54:47.329895 [DEBUG] switch_core_state_machine.c:454 (skypopen/skype101/echo123) Callstate Change HANGUP -> DOWN
2011-06-19 12:54:47.329895 [DEBUG] switch_core_state_machine.c:457 (skypopen/skype101/echo123) Running State Change CS_DESTROY
2011-06-19 12:54:47.329895 [DEBUG] switch_core_state_machine.c:467 (skypopen/skype101/echo123) State DESTROY
2011-06-19 12:54:47.329895 [DEBUG] mod_skypopen.c:480          [ac7ca63|9350fb9] [DEBUG_SKYPE  480  ][skype101       ][DOWN,INC_HNG] skype101 CHANNEL DESTROY 3cc797ba-7552-459c-9a86-9daa8745821c
2011-06-19 12:54:47.329895 [DEBUG] mod_skypopen.c:494          [ac7ca63|9350fb9] [DEBUG_SKYPE  494  ][skype101       ][DOWN,INC_HNG] audio tcp threads to DIE
2011-06-19 12:54:47.329895 [DEBUG] mod_skypopen.c:504          [ac7ca63|9350fb9] [DEBUG_SKYPE  504  ][skype101       ][DOWN,INC_HNG] audio tcp srv thread DEAD 0
2011-06-19 12:54:47.329895 [DEBUG] mod_skypopen.c:514          [ac7ca63|9350fb9] [DEBUG_SKYPE  514  ][skype101       ][DOWN,INC_HNG] audio tcp cli thread DEAD 0
2011-06-19 12:54:47.329895 [DEBUG] mod_skypopen.c:525          [ac7ca63|9350fb9] [DEBUG_SKYPE  525  ][skype101       ][DOWN,INC_HNG] codecs DOWN
2011-06-19 12:54:47.329895 [DEBUG] mod_skypopen.c:580          [ac7ca63|9350fb9] [DEBUG_SKYPE  580  ][skype101       ][IDLE,IDLE] CHANNEL DESTROYED 3cc797ba-7552-459c-9a86-9daa8745821c
2011-06-19 12:54:47.329895 [DEBUG] switch_core_state_machine.c:60 skypopen/skype101/echo123 Standard DESTROY
2011-06-19 12:54:47.329895 [DEBUG] switch_core_state_machine.c:467 (skypopen/skype101/echo123) State DESTROY going to sleep
2011-06-19 12:54:47.329895 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][IDLE,IDLE] READING: |||ERROR 68||| 
2011-06-19 12:54:47.329895 [DEBUG] skypopen_protocol.c:186     [ac7ca63|9350fb9] [DEBUG_SKYPE  186  ][skype101       ][IDLE,IDLE] If I don't connect immediately, please give the Skype client authorization to be connected by Skypopen (and to not ask you again)
2011-06-19 12:54:48.330039 [DEBUG] skypopen_protocol.c:1619    [ac7ca63|9350fb9] [DEBUG_SKYPE  1619 ][skype101       ][IDLE,IDLE] SENDING: |||PROTOCOL 7||||
2011-06-19 12:54:48.330039 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][IDLE,IDLE] READING: |||ERROR 68||| 
2011-06-19 12:54:48.330039 [DEBUG] skypopen_protocol.c:186     [ac7ca63|9350fb9] [DEBUG_SKYPE  186  ][skype101       ][IDLE,IDLE] If I don't connect immediately, please give the Skype client authorization to be connected by Skypopen (and to not ask you again)
2011-06-19 12:54:49.335187 [DEBUG] skypopen_protocol.c:1619    [ac7ca63|9350fb9] [DEBUG_SKYPE  1619 ][skype101       ][IDLE,IDLE] SENDING: |||PROTOCOL 7||||
2011-06-19 12:54:49.355190 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][IDLE,IDLE] READING: |||ERROR 68||| 
2011-06-19 12:54:49.355190 [DEBUG] skypopen_protocol.c:186     [ac7ca63|9350fb9] [DEBUG_SKYPE  186  ][skype101       ][IDLE,IDLE] If I don't connect immediately, please give the Skype client authorization to be connected by Skypopen (and to not ask you again)
2011-06-19 12:54:50.350535 [DEBUG] skypopen_protocol.c:1619    [ac7ca63|9350fb9] [DEBUG_SKYPE  1619 ][skype101       ][IDLE,IDLE] SENDING: |||PROTOCOL 7||||
2011-06-19 12:54:50.350535 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][IDLE,IDLE] READING: |||ERROR 68||| 
2011-06-19 12:54:50.350535 [DEBUG] skypopen_protocol.c:186     [ac7ca63|9350fb9] [DEBUG_SKYPE  186  ][skype101       ][IDLE,IDLE] If I don't connect immediately, please give the Skype client authorization to be connected by Skypopen (and to not ask you again)
2011-06-19 12:54:51.350481 [DEBUG] skypopen_protocol.c:1619    [ac7ca63|9350fb9] [DEBUG_SKYPE  1619 ][skype101       ][IDLE,IDLE] SENDING: |||PROTOCOL 7||||
2011-06-19 12:54:51.370489 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][IDLE,IDLE] READING: |||ERROR 68||| 
2011-06-19 12:54:51.370489 [DEBUG] skypopen_protocol.c:186     [ac7ca63|9350fb9] [DEBUG_SKYPE  186  ][skype101       ][IDLE,IDLE] If I don't connect immediately, please give the Skype client authorization to be connected by Skypopen (and to not ask you again)
2011-06-19 12:54:52.370630 [DEBUG] skypopen_protocol.c:1619    [ac7ca63|9350fb9] [DEBUG_SKYPE  1619 ][skype101       ][IDLE,IDLE] SENDING: |||PROTOCOL 7||||
2011-06-19 12:54:52.370630 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][IDLE,IDLE] READING: |||ERROR 68||| 
2011-06-19 12:54:52.370630 [DEBUG] skypopen_protocol.c:186     [ac7ca63|9350fb9] [DEBUG_SKYPE  186  ][skype101       ][IDLE,IDLE] If I don't connect immediately, please give the Skype client authorization to be connected by Skypopen (and to not ask you again)
2011-06-19 12:54:53.370777 [DEBUG] skypopen_protocol.c:1619    [ac7ca63|9350fb9] [DEBUG_SKYPE  1619 ][skype101       ][IDLE,IDLE] SENDING: |||PROTOCOL 7||||
2011-06-19 12:54:53.390783 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][IDLE,IDLE] READING: |||ERROR 68||| 
2011-06-19 12:54:53.390783 [DEBUG] skypopen_protocol.c:186     [ac7ca63|9350fb9] [DEBUG_SKYPE  186  ][skype101       ][IDLE,IDLE] If I don't connect immediately, please give the Skype client authorization to be connected by Skypopen (and to not ask you again)
2011-06-19 12:54:54.390928 [DEBUG] skypopen_protocol.c:1619    [ac7ca63|9350fb9] [DEBUG_SKYPE  1619 ][skype101       ][IDLE,IDLE] SENDING: |||PROTOCOL 7||||
2011-06-19 12:54:54.390928 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][IDLE,IDLE] READING: |||ERROR 68||| 
2011-06-19 12:54:54.390928 [DEBUG] skypopen_protocol.c:186     [ac7ca63|9350fb9] [DEBUG_SKYPE  186  ][skype101       ][IDLE,IDLE] If I don't connect immediately, please give the Skype client authorization to be connected by Skypopen (and to not ask you again)
2011-06-19 12:54:55.392072 [DEBUG] skypopen_protocol.c:1619    [ac7ca63|9350fb9] [DEBUG_SKYPE  1619 ][skype101       ][IDLE,IDLE] SENDING: |||PROTOCOL 7||||
2011-06-19 12:54:55.412076 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][IDLE,IDLE] READING: |||ERROR 68||| 
2011-06-19 12:54:55.412076 [DEBUG] skypopen_protocol.c:186     [ac7ca63|9350fb9] [DEBUG_SKYPE  186  ][skype101       ][IDLE,IDLE] If I don't connect immediately, please give the Skype client authorization to be connected by Skypopen (and to not ask you again)
2011-06-19 12:54:56.407220 [DEBUG] skypopen_protocol.c:1619    [ac7ca63|9350fb9] [DEBUG_SKYPE  1619 ][skype101       ][IDLE,IDLE] SENDING: |||PROTOCOL 7||||
2011-06-19 12:54:56.428426 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][IDLE,IDLE] READING: |||ERROR 68||| 
2011-06-19 12:54:56.428426 [DEBUG] skypopen_protocol.c:186     [ac7ca63|9350fb9] [DEBUG_SKYPE  186  ][skype101       ][IDLE,IDLE] If I don't connect immediately, please give the Skype client authorization to be connected by Skypopen (and to not ask you again)
2011-06-19 12:54:57.430371 [DEBUG] skypopen_protocol.c:1619    [ac7ca63|9350fb9] [DEBUG_SKYPE  1619 ][skype101       ][IDLE,IDLE] SENDING: |||PROTOCOL 7||||
2011-06-19 12:54:57.430371 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][IDLE,IDLE] READING: |||ERROR 68||| 
2011-06-19 12:54:57.430371 [DEBUG] skypopen_protocol.c:186     [ac7ca63|9350fb9] [DEBUG_SKYPE  186  ][skype101       ][IDLE,IDLE] If I don't connect immediately, please give the Skype client authorization to be connected by Skypopen (and to not ask you again)
2011-06-19 12:54:58.430517 [DEBUG] skypopen_protocol.c:1619    [ac7ca63|9350fb9] [DEBUG_SKYPE  1619 ][skype101       ][IDLE,IDLE] SENDING: |||PROTOCOL 7||||
2011-06-19 12:54:58.450520 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][IDLE,IDLE] READING: |||ERROR 68||| 
2011-06-19 12:54:58.450520 [DEBUG] skypopen_protocol.c:186     [ac7ca63|9350fb9] [DEBUG_SKYPE  186  ][skype101       ][IDLE,IDLE] If I don't connect immediately, please give the Skype client authorization to be connected by Skypopen (and to not ask you again)
2011-06-19 12:54:59.450666 [DEBUG] skypopen_protocol.c:1619    [ac7ca63|9350fb9] [DEBUG_SKYPE  1619 ][skype101       ][IDLE,IDLE] SENDING: |||PROTOCOL 7||||
2011-06-19 12:54:59.450666 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][IDLE,IDLE] READING: |||ERROR 68||| 
2011-06-19 12:54:59.450666 [DEBUG] skypopen_protocol.c:186     [ac7ca63|9350fb9] [DEBUG_SKYPE  186  ][skype101       ][IDLE,IDLE] If I don't connect immediately, please give the Skype client authorization to be connected by Skypopen (and to not ask you again)
2011-06-19 12:55:00.450812 [DEBUG] skypopen_protocol.c:1619    [ac7ca63|9350fb9] [DEBUG_SKYPE  1619 ][skype101       ][IDLE,IDLE] SENDING: |||PROTOCOL 7||||
2011-06-19 12:55:00.470814 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][IDLE,IDLE] READING: |||ERROR 68||| 
2011-06-19 12:55:00.470814 [DEBUG] skypopen_protocol.c:186     [ac7ca63|9350fb9] [DEBUG_SKYPE  186  ][skype101       ][IDLE,IDLE] If I don't connect immediately, please give the Skype client authorization to be connected by Skypopen (and to not ask you again)
2011-06-19 12:55:01.451959 [DEBUG] skypopen_protocol.c:1619    [ac7ca63|9350fb9] [DEBUG_SKYPE  1619 ][skype101       ][IDLE,IDLE] SENDING: |||PROTOCOL 7||||
2011-06-19 12:55:01.471961 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][IDLE,IDLE] READING: |||ERROR 68||| 
2011-06-19 12:55:01.471961 [DEBUG] skypopen_protocol.c:186     [ac7ca63|9350fb9] [DEBUG_SKYPE  186  ][skype101       ][IDLE,IDLE] If I don't connect immediately, please give the Skype client authorization to be connected by Skypopen (and to not ask you again)
2011-06-19 12:55:02.479109 [DEBUG] skypopen_protocol.c:1619    [ac7ca63|9350fb9] [DEBUG_SKYPE  1619 ][skype101       ][IDLE,IDLE] SENDING: |||PROTOCOL 7||||
2011-06-19 12:55:02.479109 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][IDLE,IDLE] READING: |||ERROR 68||| 
2011-06-19 12:55:02.479109 [DEBUG] skypopen_protocol.c:186     [ac7ca63|9350fb9] [DEBUG_SKYPE  186  ][skype101       ][IDLE,IDLE] If I don't connect immediately, please give the Skype client authorization to be connected by Skypopen (and to not ask you again)
2011-06-19 12:55:03.492257 [DEBUG] skypopen_protocol.c:1619    [ac7ca63|9350fb9] [DEBUG_SKYPE  1619 ][skype101       ][IDLE,IDLE] SENDING: |||PROTOCOL 7||||
2011-06-19 12:55:03.492257 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][IDLE,IDLE] READING: |||ERROR 68||| 
2011-06-19 12:55:03.492257 [DEBUG] skypopen_protocol.c:186     [ac7ca63|9350fb9] [DEBUG_SKYPE  186  ][skype101       ][IDLE,IDLE] If I don't connect immediately, please give the Skype client authorization to be connected by Skypopen (and to not ask you again)
2011-06-19 12:55:04.495411 [DEBUG] skypopen_protocol.c:1619    [ac7ca63|9350fb9] [DEBUG_SKYPE  1619 ][skype101       ][IDLE,IDLE] SENDING: |||PROTOCOL 7||||
2011-06-19 12:55:04.515407 [DEBUG] skypopen_protocol.c:173     [ac7ca63|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][IDLE,IDLE] READING: |||ERROR 68||| 
2011-06-19 12:55:04.515407 [DEBUG] skypopen_protocol.c:186     [ac7ca63|9350fb9] [DEBUG_SKYPE  186  ][skype101       ][IDLE,IDLE] If I don't connect immediately, please give the Skype client authorization to be connected by Skypopen (and to not ask you again)

From bryan.lemon.iccle at gmail.com  Mon Jun 20 00:47:20 2011
From: bryan.lemon.iccle at gmail.com (Bryan Lemon)
Date: Sun, 19 Jun 2011 16:47:20 -0400
Subject: [Freeswitch-users] Loading Pocketsphinx
Message-ID: <BANLkTimjN9=4DH57c9fk37UKKEu8aZ=emw@mail.gmail.com>

Whenever I try to load the pocketsphinx module, either on startup, or with
the load mod_pocketsphinx command, I get the following error:
freeswitch at bryanlemon-laptop> load mod_pocketsphinx
2011-06-19 15:17:04.471275 [INFO] mod_enum.c:775 ENUM Reloaded
2011-06-19 15:17:04.471275 [INFO] switch_time.c:1020 Timezone reloaded 530
definitions

+OK Reloading XML
-ERR [module load file routine returned an error]

2011-06-19 15:17:04.471275 [CRIT] switch_loadable_module.c:928 Error Loading
module /usr/local/freeswitch/mod/mod_pocketsphinx.so
**/usr/local/freeswitch/mod/mod_pocketsphinx.so: undefined symbol:
feat_init**


I am running Ubuntu 10.04, and installed freeswitch using the git method. I
tried a git pull && make current, and it runs through with no problem.

Any Ideas?

Thank you,
Bryan Lemon
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From endlesspixel at hotmail.com  Mon Jun 20 06:24:49 2011
From: endlesspixel at hotmail.com (J.B. BERLIN)
Date: Mon, 20 Jun 2011 04:24:49 +0200
Subject: [Freeswitch-users] Have someone zrtp with correct sas ?
Message-ID: <BAY164-ds10EADF64CEECCFCDCF0F29C16E0@phx.gbl>

Dear users,

I try run FS last git build from today 19. June 2011 with ZRTP.

Zfone show me green but the sas are everytime different!

Tested with 2x Notebooks on Fedora 14 Twinkle(via yum) & with two desktops with Win7 x64 / Zfone 0.92.build.219 and X-Lite on both sides.

FS is build with libzrtp-0.81.514

zrtp.lua is enabled !

global_setvar zrtp_secure_media=true" is set

Or have someone a real good step by step guide ?

Many thanks and best regards
Jacek
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From a.luppi at seletech.com  Mon Jun 20 19:04:03 2011
From: a.luppi at seletech.com (Alessandro)
Date: Mon, 20 Jun 2011 17:04:03 +0200
Subject: [Freeswitch-users] Call Hangup
In-Reply-To: <BANLkTinxDahbHqi5ACUev5cyj_b=sWR+tQ@mail.gmail.com>
References: <4DFEFCE0.9040809@seletech.com>	<B8D314BD-A38C-4B6D-AC95-84643B433E9C@ipeva.fr>	<4DFF12B8.5090008@seletech.com>
	<BANLkTinxDahbHqi5ACUev5cyj_b=sWR+tQ@mail.gmail.com>
Message-ID: <4DFF6163.7010901@seletech.com>

Yes, i have done only reload xml with fusionpbx.

Thanks

Alessandro

Il 20/06/2011 16:33, Avi Marcus ha scritto:
> You need to do a "sofia profile internal restart" to load your new 
> settings.
>
> -Avi Marcus
>
>
> On Mon, Jun 20, 2011 at 12:28 PM, Alessandro <a.luppi at seletech.com 
> <mailto:a.luppi at seletech.com>> wrote:
>
>     Hi,
>
>     i set
>     <param name="enable-timer" value="true"/>
>     <param name="session-timeout" value="120"/>
>
>     on sip profile, but doesn't change. FS doesn't hang up the call
>     after one of the two extension disappear from the network.
>     With wireshark i don't see re-invite during the call.
>
>     Regards
>
>     Alessandro
>      Il 20/06/2011 10:43, David Ponzone ha scritto:
>>     Alessandro,
>>
>>     there are 2 ways to do that:
>>     Session timers: with this, FS will send a RE-INVITE to phones
>>     every X minutes, and if the phone does not answer, it's
>>     considerer dead, so it is unregistered and the current calls will
>>     be hung up
>>     or
>>     RTP Timeout: this will detect that media is missing, and will
>>     hang up the call
>>
>>     Session timers has to be enabled on the SIP Profile with:
>>     <param name="enable-timer" value="true"/>
>>     <param name="session-timeout" value="120"/>
>>
>>     RTP timeout has to be enabled in the SIP Profile with:
>>     <param name="rtp-timeout-sec" value="300"/>
>>     (use the value you wish. Also, be careful with this, it can lead
>>     to issues with some phones when you put a call on hold/mute or if
>>     you use VAD)
>>
>>     David Ponzone Direction Technique
>>     email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>     tel:      01 74 03 18 97
>>     gsm:   06 66 98 76 34
>>
>>     Service ClientIPeva
>>     tel:      0811 46 26 26
>>     www.ipeva.fr  - www.ipeva-studio.com
>>
>>     /Ce message et toutes les pi?ces jointes sont confidentiels et
>>     ?tablis ? l'intention exclusive de ses destinataires. Toute
>>     utilisation ou diffusion non autoris?e est interdite. Tout
>>     message ?lectronique est susceptible d'alt?ration.
>>     /*/IPeva/*/ d?cline toute responsabilit? au titre de ce message
>>     s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>     destinataire de ce message, merci de le d?truire imm?diatement et
>>     d'avertir l'exp?diteur./
>>     /
>>     /
>>
>>
>>
>>     Le 20/06/2011 ? 09:55, Alessandro a ?crit :
>>
>>>     Hi,
>>>
>>>     I have an active call from the extension A to the extension B.
>>>     If one of the two extension is disconnected from the network
>>>     without
>>>     deregistration the call isn't hung up by the server. The call
>>>     should be
>>>     hangup when FS realizes that the extension isn't in the network?
>>>     Is there a time out to be set somewhere?
>>>
>>>     thanks
>>>
>>>     Regards
>>>
>>>     Alessandro Luppi
>>>
>>>     -- 
>>>     Ing. Alessandro Luppi
>>>     Software development
>>>     Seletech srl
>>>     Via Collodi 8, 20052 Monza (MI) - Italy
>>>     Tel: +39.039.5962000 - Fax: +39.039.9716905
>>>     email: a.luppi at seletech.com <mailto:a.luppi at seletech.com> - Web:
>>>     www.seletech.com <http://www.seletech.com>   or www.seletech.eu
>>>     <http://www.seletech.eu>
>>>
>>>
>>>     _______________________________________________
>>>     Join us at ClueCon 2011, Aug 9-11, Chicago
>>>     http://www.cluecon.com 877-7-4ACLUE
>>>
>>>     FreeSWITCH-users mailing list
>>>     FreeSWITCH-users at lists.freeswitch.org
>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>
>>
>>     _______________________________________________
>>     Join us at ClueCon 2011, Aug 9-11, Chicago
>>     http://www.cluecon.com  877-7-4ACLUE
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>     -- 
>     Ing. Alessandro Luppi
>     Software development
>     Seletech srl
>     Via Collodi 8, 20052 Monza (MI) - Italy
>     Tel: +39.039.5962000 - Fax: +39.039.9716905
>     email:a.luppi at seletech.com  <mailto:a.luppi at seletech.com>  - Web:www.seletech.com  <http://www.seletech.com>    orwww.seletech.eu  <http://www.seletech.eu>
>
>
>     _______________________________________________
>     Join us at ClueCon 2011, Aug 9-11, Chicago
>     http://www.cluecon.com 877-7-4ACLUE
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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

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From avi at avimarcus.net  Mon Jun 20 19:12:14 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 20 Jun 2011 18:12:14 +0300
Subject: [Freeswitch-users] Call Hangup
In-Reply-To: <4DFF6163.7010901@seletech.com>
References: <4DFEFCE0.9040809@seletech.com>
	<B8D314BD-A38C-4B6D-AC95-84643B433E9C@ipeva.fr>
	<4DFF12B8.5090008@seletech.com>
	<BANLkTinxDahbHqi5ACUev5cyj_b=sWR+tQ@mail.gmail.com>
	<4DFF6163.7010901@seletech.com>
Message-ID: <BANLkTim3tJ7EU1EDynVp6UAorUtE5Yj3hw@mail.gmail.com>

BTW, fusionpbx on the Status -> SIP Status page has buttons to
start/stop/restart the profiles.

 -Avi Marcus


On Mon, Jun 20, 2011 at 6:04 PM, Alessandro <a.luppi at seletech.com> wrote:

> **
> Yes, i have done only reload xml with fusionpbx.
>
> Thanks
>
> Alessandro
>
> Il 20/06/2011 16:33, Avi Marcus ha scritto:
>
> You need to do a "sofia profile internal restart" to load your new
> settings.
>
>   -Avi Marcus
>
> On Mon, Jun 20, 2011 at 12:28 PM, Alessandro <a.luppi at seletech.com> wrote:
>
>>  Hi,
>>
>> i set
>>  <param name="enable-timer" value="true"/>
>> <param name="session-timeout" value="120"/>
>>
>> on sip profile, but doesn't change. FS doesn't hang up the call after one
>> of the two extension disappear from the network.
>> With wireshark i don't see re-invite during the call.
>>
>> Regards
>>
>> Alessandro
>>   Il 20/06/2011 10:43, David Ponzone ha scritto:
>>
>> Alessandro,
>>
>>  there are 2 ways to do that:
>> Session timers: with this, FS will send a RE-INVITE to phones every X
>> minutes, and if the phone does not answer, it's considerer dead, so it is
>> unregistered and the current calls will be hung up
>> or
>> RTP Timeout: this will detect that media is missing, and will hang up the
>> call
>>
>>  Session timers has to be enabled on the SIP Profile with:
>> <param name="enable-timer" value="true"/>
>> <param name="session-timeout" value="120"/>
>>
>>  RTP timeout has to be enabled in the SIP Profile with:
>>       <param name="rtp-timeout-sec" value="300"/>
>>  (use the value you wish. Also, be careful with this, it can lead to
>> issues with some phones when you put a call on hold/mute or if you use VAD)
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>>  Service Client IPeva
>>  tel:      0811 46 26 26
>>  www.ipeva.fr  -   www.ipeva-studio.com
>>
>>  *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>>  Le 20/06/2011 ? 09:55, Alessandro a ?crit :
>>
>>  Hi,
>>
>> I have an active call from the extension A to the extension B.
>> If one of the two extension is disconnected from the network without
>> deregistration the call isn't hung up by the server. The call should be
>> hangup when FS realizes that the extension isn't in the network?
>> Is there a time out to be set somewhere?
>>
>> thanks
>>
>> Regards
>>
>> Alessandro Luppi
>>
>> --
>> Ing. Alessandro Luppi
>> Software development
>> Seletech srl
>> Via Collodi 8, 20052 Monza (MI) - Italy
>> Tel: +39.039.5962000 - Fax: +39.039.9716905
>> email: a.luppi at seletech.com - Web: www.seletech.com   or
>> www.seletech.eu
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> --
>> Ing. Alessandro Luppi
>> Software development
>> Seletech srl
>> Via Collodi 8, 20052 Monza (MI) - Italy
>> Tel: +39.039.5962000 - Fax: +39.039.9716905
>> email: a.luppi at seletech.com - Web: www.seletech.com   or   www.seletech.eu
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> Ing. Alessandro Luppi
> Software development
> Seletech srl
> Via Collodi 8, 20052 Monza (MI) - Italy
> Tel: +39.039.5962000 - Fax: +39.039.9716905
> email: a.luppi at seletech.com - Web: www.seletech.com   or   www.seletech.eu
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jan.berger at video24.no  Mon Jun 20 19:22:21 2011
From: jan.berger at video24.no (Jan Berger)
Date: Mon, 20 Jun 2011 17:22:21 +0200
Subject: [Freeswitch-users] Loading Pocketsphinx
In-Reply-To: <BANLkTimjN9=4DH57c9fk37UKKEu8aZ=emw@mail.gmail.com>
References: <BANLkTimjN9=4DH57c9fk37UKKEu8aZ=emw@mail.gmail.com>
Message-ID: <3277E95E06DC465F885DD89B0FFB7355@dell9400>

You can also use pocketsphinx through uniMRCP according to their website. I
have not tried it myself yet but I will a bit later.

 

Jan

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bryan
Lemon
Sent: 19. juni 2011 22:47
To: freeswitch-users
Subject: [Freeswitch-users] Loading Pocketsphinx

 

Whenever I try to load the pocketsphinx module, either on startup, or with
the load mod_pocketsphinx command, I get the following error:

freeswitch at bryanlemon-laptop> load mod_pocketsphinx

2011-06-19 15:17:04.471275 [INFO] mod_enum.c:775 ENUM Reloaded

2011-06-19 15:17:04.471275 [INFO] switch_time.c:1020 Timezone reloaded 530
definitions

 

+OK Reloading XML

-ERR [module load file routine returned an error]

 

2011-06-19 15:17:04.471275 [CRIT] switch_loadable_module.c:928 Error Loading
module /usr/local/freeswitch/mod/mod_pocketsphinx.so

**/usr/local/freeswitch/mod/mod_pocketsphinx.so: undefined symbol:
feat_init**

 

 

I am running Ubuntu 10.04, and installed freeswitch using the git method. I
tried a git pull && make current, and it runs through with no problem. 

 

Any Ideas?

Thank you,
Bryan Lemon

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From a.luppi at seletech.com  Mon Jun 20 19:26:08 2011
From: a.luppi at seletech.com (Alessandro)
Date: Mon, 20 Jun 2011 17:26:08 +0200
Subject: [Freeswitch-users] Call Hangup
In-Reply-To: <BANLkTim3tJ7EU1EDynVp6UAorUtE5Yj3hw@mail.gmail.com>
References: <4DFEFCE0.9040809@seletech.com>	<B8D314BD-A38C-4B6D-AC95-84643B433E9C@ipeva.fr>	<4DFF12B8.5090008@seletech.com>	<BANLkTinxDahbHqi5ACUev5cyj_b=sWR+tQ@mail.gmail.com>	<4DFF6163.7010901@seletech.com>
	<BANLkTim3tJ7EU1EDynVp6UAorUtE5Yj3hw@mail.gmail.com>
Message-ID: <4DFF6690.3020804@seletech.com>

yes,

done.. My mistake I thought that was enough reload xml

Thanks for help

Alessandro

Il 20/06/2011 17:12, Avi Marcus ha scritto:
> IP Status page has buttons to start/stop/restart the profiles.
>


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



From msc at freeswitch.org  Mon Jun 20 19:26:47 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 20 Jun 2011 08:26:47 -0700
Subject: [Freeswitch-users] FreeSWITCH Cookbook Recipe List
In-Reply-To: <7EAFE942-322C-4EC0-941A-F1226B8DDAD1@mgtech.com>
References: <BANLkTima7xQPi2JYe908CfyZaYmOMsw=aQ@mail.gmail.com>
	<7EAFE942-322C-4EC0-941A-F1226B8DDAD1@mgtech.com>
Message-ID: <BANLkTinuMMORNRSZPbF0MVVTUXugpMRCNA@mail.gmail.com>

I believe we decided to forego the "how-to install fs" recipes since we have
lots and lots of information on that subject on the wiki. We want to keep
the recipes focused on interesting ways to use FreeSWITCH.

-MC

On Mon, Jun 20, 2011 at 7:55 AM, Mario G <mario_fs at mgtech.com> wrote:

> I see there is a need for install on OS X Server. I thought the OS X
> install I did for the wiki was quite comprehensive. What is missing? It
> should be the same for OS X server. Also, I will do a clean install on OS X
> 10.7 next month to update the wiki.
>
> On Jun 17, 2011, at 8:14 PM, Michael Collins wrote:
>
> Hello all!
>
> Some have inquired recently about what's needed to complete the FS
> cookbook. I've uploaded the most recent list of recipes that we have
> identified. Some are already written but many are yet to be done. Check out
> this page:
>
>
> http://wiki.freeswitch.org/wiki/Cookbook#Information_for_prospective_authors
>
> Contact me if you have any questions about helping with any of these
> recipes.
>
> Thanks,
> Michael
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Jun 20 19:31:25 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 20 Jun 2011 08:31:25 -0700
Subject: [Freeswitch-users] FreeSWITCH Cookbook Recipe List
In-Reply-To: <BANLkTimn9OzKORpuG259rTRFGPwdLbd9FA@mail.gmail.com>
References: <BANLkTimn9OzKORpuG259rTRFGPwdLbd9FA@mail.gmail.com>
Message-ID: <BANLkTinJS-+YtZugOgxSoaH3h=yeRWYPjA@mail.gmail.com>

Mitch,

if you can supply how-to's/examples/screen shots for SRTP/TLS and various
phones then those can make it into the chapter on security. We do mention
setting up TLS and SRTP but like you said, each phone is different. A few
examples will help.

-MC

On Mon, Jun 20, 2011 at 6:24 AM, mitch Johnson <mitch.johnson7 at gmail.com>wrote:

> It seems as if you have already identified what's going into the Cookbook.
> Is it too late to add a few sections on TLS/SRTP with examples on using
> different phone vendors and softphones on smartphones?
>
> I think FreeSWITCH is a powerful solution in the TLS/SRTP area yet to be
> realized or tapped into.  Having recipes on how to implement and a great
> discussion on how FreeSWITCH can use channels on a single box to do
> encrypted calls on the Internet side and unencrypted calls on the LAN side
> would be incredible.
>
> Thanks,
>
> Mitch Johnson
>
>>
>> ---------- Forwarded message ----------
>> From: Michael Collins <msc at freeswitch.org>
>> To: freeswitch-users at lists.freeswitch.org,
>> freeswitch-dev at lists.freeswitch.org
>> Date: Fri, 17 Jun 2011 20:14:33 -0700
>> Subject: [Freeswitch-users] FreeSWITCH Cookbook Recipe List
>> Hello all!
>>
>> Some have inquired recently about what's needed to complete the FS
>> cookbook. I've uploaded the most recent list of recipes that we have
>> identified. Some are already written but many are yet to be done. Check out
>> this page:
>>
>>
>> http://wiki.freeswitch.org/wiki/Cookbook#Information_for_prospective_authors
>>
>> Contact me if you have any questions about helping with any of these
>> recipes.
>>
>> Thanks,
>> Michael
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Mon Jun 20 19:39:21 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 20 Jun 2011 10:39:21 -0500
Subject: [Freeswitch-users] FreeSWITCH Cookbook Recipe List
In-Reply-To: <BANLkTinJS-+YtZugOgxSoaH3h=yeRWYPjA@mail.gmail.com>
References: <BANLkTimn9OzKORpuG259rTRFGPwdLbd9FA@mail.gmail.com>
	<BANLkTinJS-+YtZugOgxSoaH3h=yeRWYPjA@mail.gmail.com>
Message-ID: <69DAAD6A-4597-4CFF-8D57-08CB6DA5A147@freeswitch.org>

This is documented really well on the Wiki and is tested a lot by myself and I think Mitch Capper?

/b

On Jun 20, 2011, at 10:31 AM, Michael Collins wrote:

> Mitch,
> 
> if you can supply how-to's/examples/screen shots for SRTP/TLS and various
> phones then those can make it into the chapter on security. We do mention
> setting up TLS and SRTP but like you said, each phone is different. A few
> examples will help.
> 
> -MC



From anton.vazir at gmail.com  Mon Jun 20 20:21:40 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Mon, 20 Jun 2011 21:21:40 +0500
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <84b3f6b0-9cff-40cc-8a32-f1a2da1458ab@zimbra1.crocker.com>
References: <0013F8A2-8D82-470C-B8D8-45C22BDB3A00@ipeva.fr>
	<84b3f6b0-9cff-40cc-8a32-f1a2da1458ab@zimbra1.crocker.com>
Message-ID: <BANLkTimJWo21Pi4301-BCCDZk3AuFjzjKA@mail.gmail.com>

zyxel have what you ask. Huawei have better pricing (~20$ per port)
but huawei wants to sell huge amounts and quality is somehow bad.

Zyxel is quite good (did not try ZyXEL Voip features, only ADSL) but
ADSL is good and reliable, but more expensive per port. There are
options for 12/24/48 and modular 300+ ports

2011/6/20 Matthew S. Crocker <matthew at corp.crocker.com>:
>
> Zhone MALC-XP are nowhere near $11k.??? They are < $100/port
>
> ________________________________
>
> From: "David Ponzone" <david.ponzone at ipeva.fr>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Friday, June 17, 2011 5:29:48 AM
> Subject: Re: [Freeswitch-users] VoIP IP DSLAMs
>
> Hmmm unlikely I think.
> The Zhone box for instance seems to be around 11,000$.
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 17/06/2011 ? 10:25, Nandy Dagondon a ?crit :
>
> yes, it doesn't add up because the application is in a rural area. the
> Phybridge is a good solution except for the price. thanks David. however,
> the MSAN, perfectly describes what i'm looking - hopefully the investment is
> lower.
> thanks to all who contributed.
>
> On Fri, Jun 17, 2011 at 3:37 PM, David Ponzone <david.ponzone at ipeva.fr>
> wrote:
>>
>> Nandy,
>> something does not add up.
>> When you deploy DSLAMs for public lines, it's not for 24/48 ports, but
>> rather for 20 000 to 30 000 ports.
>> Perhaps there is a confusion about what is a public line.
>> A public line is the 200 meters-8km pair of copper going from the local
>> (legacy) telco facility to your house.
>> You don't install your own DSLAMs for 24/48 ports, or perhaps you are in a
>> very specific situation.
>> I really doubt you will find a cost effective DSLAM for so few ports.
>> Plus, you realize you will need to install on DSLAM in all the telco
>> facilities required to get the coverage you need
>> There are small DSLAMs on the market, but they are targeted for private
>> lines (hospitality, health care, ...) and I really don't know if they can
>> work on public lines.
>> In case you actually need a small DSLAM just to run ADSL over a private
>> pair of copper, you should have a look at:
>> http://www.phybridge.com/uniphyer-ip-phones.aspx
>> This thing does not call itself a DSLAM but it is one.
>> They market that as a "switch" that enables LAN on your copper wires.
>> It's a small lie for data usage as you only get 25Mbps/1.4Mbps (ATM), but
>> for voice, that's enough.
>> Basically, you put the "switch"-DSLAM in the network cabinet where the
>> copper wires go, you connect the wires to it, you connect the ethernet port
>> to your LAN (so your FS). In each room/office, you plug an adapter (provided
>> also by Phybridge) which is basically a small ADSL modem (powered from the
>> switch through the wire), from which you get an ethernet port, with POE on
>> it!
>> I like the solution on the paper, but never used. Because of the cost:
>> a 24 ports switch is 3500$
>> one adapter is 122$
>> so a full 24 ports solution would be 6500$, which is not very competitive
>> compared to recabling the place.
>> But your mileage may vary.
>> David Ponzone ?Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel: ? ? ?01 74 03 18 97
>> gsm: ? 06 66 98 76 34
>> Service Client?IPeva
>> tel: ? ? ?0811 46 26 26
>> www.ipeva.fr? -? ?www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>> Le 17/06/2011 ? 08:28, Nandy Dagondon a ?crit :
>>
>> yes, without ATM layer and it's for public lines.
>>
>>
>>
>> On Fri, Jun 17, 2011 at 2:19 PM, David Ponzone <david.ponzone at ipeva.fr>
>> wrote:
>>>
>>> Which density are you looking for ?
>>> Is it for use in a private building or on public lines ?
>>> David Ponzone ?Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel: ? ? ?01 74 03 18 97
>>> gsm: ? 06 66 98 76 34
>>> Service Client?IPeva
>>> tel: ? ? ?0811 46 26 26
>>> www.ipeva.fr? -? ?www.ipeva-studio.com
>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>
>>>
>>>
>>> Le 17/06/2011 ? 08:07, Nandy Dagondon a ?crit :
>>>
>>> hello guys!
>>>
>>> i'm trying to setup a small exchange w/ Internet service - of course,
>>> using FS as the exchange. DSLAMs connect to POTS via 2-wire splitters and
>>> that would require using FXS gateway (as what Rob mentioned).
>>>
>>> i'm trying to find out if there's a way we can omit the FXS - so it's
>>> FS-to-DSLAM direct via IP.? i'm not sure if this setup is now possible on
>>> newer IP DSLAMs.
>>>
>>>
>>> On Fri, Jun 17, 2011 at 10:45 AM, jay binks <jaybinks at gmail.com> wrote:
>>>>
>>>> this is MY take on what Nandy is after.
>>>> A Dslam provides DSL on certain frequencies of the line..
>>>> ( some of ) the other frequencies are used for voice.
>>>> my understanding is that sometimes this is split off to another device
>>>> to provide the voice,
>>>> but in this case Nandy is after a DSlam that will do the DSL part AND
>>>> the voice frequencies ( Voice signalling / audio by SIP / RTP )
>>>> I guess this is a logical question and would be quite interesting to see
>>>> if there are such devices..
>>>> sorry I dont know of any :(
>>>> Jay
>>>>
>>>> On Fri, Jun 17, 2011 at 11:31 AM, Rob Hutton <justlikeef at gmail.com>
>>>> wrote:
>>>>>
>>>>> Can you explain further what you are trying to do? ?A DSLAM is a device
>>>>> that provides DSL transport, which is independent of what you run across it.
>>>>> ?So normally, you have
>>>>>
>>>>> |Server | ----|network|----|DSLAM| --------[dsl line] -------[modem or
>>>>> modem/router combo]-----[network]---[Client]
>>>>>
>>>>> What the Server is (Could be Freeswitch) and what the Client is (could
>>>>> be a VOIP phone) are independent of the transport??? ?Onewire, Cisco, and
>>>>> others make some DSL Modem/Router combos with integrated Voip to FXS ports
>>>>> for the CPE end.
>>>>>
>>>>> If you find a DSLAM running one of the OSs that Freeswitch supports,
>>>>> you could compile Freeswitch for it, but I haven't seen one with enough CPU
>>>>> tp handle much besides the traffic.
>>>>>
>>>>> On Thursday 16 June 2011 19:03:50 Nandy Dagondon wrote:
>>>>> > hello everybody,
>>>>> >
>>>>> > i'm looking for small (24/48 ports) ?IP DSLAMs that inter-connects
>>>>> > directly
>>>>> > to FreeSwitch via IP. ?is this already available on the market? or do
>>>>> > we
>>>>> > still have to connect FreeSwitch via POTS splitters and FXS gateways?
>>>>> > i
>>>>> > appreciate if you can mention some brands/models.
>>>>> >
>>>>> > tks,
>>>>> > nandy
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> --
>>>> Sincerely
>>>>
>>>> Jay
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From steveayre at gmail.com  Mon Jun 20 20:44:47 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 20 Jun 2011 17:44:47 +0100
Subject: [Freeswitch-users] Skypopen,
 Skype 2.2.0.35 Beta for Linux and Skype Connect
In-Reply-To: <BANLkTi=ru21gP4nLpSKSLqMzwBqzwaWHHQ@mail.gmail.com>
References: <BANLkTim+WjTGiTFNXBdTLqSZ_0t-DVhhmA@mail.gmail.com>
	<BANLkTi=ru21gP4nLpSKSLqMzwBqzwaWHHQ@mail.gmail.com>
Message-ID: <BANLkTimFPO4JuO0byT4i45C89vG4Teb7GA@mail.gmail.com>

I don't think that version's supported...

http://wiki.freeswitch.org/wiki/Mod_skypopen*:
"Don't* use the *beta* Skype client or more recent "stable" (2.0.0.72 static
for OSS is the one you want to use)."

Giovanni is the module maintainer so can probably chip in to clarify if
that's still the case or whether that's or information.

-Steve



On 19 June 2011 12:34, Skyp Tester <skyptester at gmail.com> wrote:

> Hi
>
>
>
> There is problem with Skype Linux client 2.0.0.72 , I cant call to Skype
> Connect client, error ?FAILUREREASON 2? = ?User or phone number does not
> exist?.
>
> http://www.skype.com/intl/en-us/business/skype-connect/
>
>
>
> Installed skypopen with Skype client 2.0.0.72 .
>
> Upgraded to 2.2.0.35:
>
> 1)      install libstdc++-4.3.2-7.i386.rpm
>
> 2)      copy over /usr/bin/skype and /usr/share/skype/lang with new
> version
>
>
>
> Now I can call to Skype Connect client but there are some problems
>
> 1)      audio is one way, Skype->FS
>
> 2)      after first call there is error ?NO SPACE WRITE: 640? and no more
> Skype calls, ?ERROR 68? & ?If I don't connect immediately, please give the
> Skype client authorization to be connected by Skypopen  (and to not ask
> you again)?
>
>
>
> Can we hope support to Skype Linux 2.2.0.35 or there is some big technical
> obstacle?
>
>
>
>
>
> Centos 5.6 x86_64
>
> 2.6.18-238.12.1.el5 #1 SMP
>
> FreeSWITCH Version 1.0.head (git-fccbba5 2011-05-18 19-00-42 -0400)
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gmaruzz at gmail.com  Mon Jun 20 20:51:21 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 20 Jun 2011 18:51:21 +0200
Subject: [Freeswitch-users] Skypopen,
 Skype 2.2.0.35 Beta for Linux and Skype Connect
In-Reply-To: <BANLkTimFPO4JuO0byT4i45C89vG4Teb7GA@mail.gmail.com>
References: <BANLkTim+WjTGiTFNXBdTLqSZ_0t-DVhhmA@mail.gmail.com>
	<BANLkTi=ru21gP4nLpSKSLqMzwBqzwaWHHQ@mail.gmail.com>
	<BANLkTimFPO4JuO0byT4i45C89vG4Teb7GA@mail.gmail.com>
Message-ID: <BANLkTimA25tKy+CCgJC6bpEYzw+Pbzs=hQ@mail.gmail.com>

All relevant info are in the wiki page ;).

Particularly, I would counseil to use the interactive installer, that
does it all automatically for you. Included downloading, configuring
and installing the correct skype client.

-giovanni


On 6/20/11, Steven Ayre <steveayre at gmail.com> wrote:
> I don't think that version's supported...
>
> http://wiki.freeswitch.org/wiki/Mod_skypopen*:
> "Don't* use the *beta* Skype client or more recent "stable" (2.0.0.72 static
> for OSS is the one you want to use)."
>
> Giovanni is the module maintainer so can probably chip in to clarify if
> that's still the case or whether that's or information.
>
> -Steve
>
>
>
> On 19 June 2011 12:34, Skyp Tester <skyptester at gmail.com> wrote:
>
>> Hi
>>
>>
>>
>> There is problem with Skype Linux client 2.0.0.72 , I cant call to Skype
>> Connect client, error ?FAILUREREASON 2? = ?User or phone number does not
>> exist?.
>>
>> http://www.skype.com/intl/en-us/business/skype-connect/
>>
>>
>>
>> Installed skypopen with Skype client 2.0.0.72 .
>>
>> Upgraded to 2.2.0.35:
>>
>> 1)      install libstdc++-4.3.2-7.i386.rpm
>>
>> 2)      copy over /usr/bin/skype and /usr/share/skype/lang with new
>> version
>>
>>
>>
>> Now I can call to Skype Connect client but there are some problems
>>
>> 1)      audio is one way, Skype->FS
>>
>> 2)      after first call there is error ?NO SPACE WRITE: 640? and no more
>> Skype calls, ?ERROR 68? & ?If I don't connect immediately, please give the
>> Skype client authorization to be connected by Skypopen  (and to not ask
>> you again)?
>>
>>
>>
>> Can we hope support to Skype Linux 2.2.0.35 or there is some big technical
>> obstacle?
>>
>>
>>
>>
>>
>> Centos 5.6 x86_64
>>
>> 2.6.18-238.12.1.el5 #1 SMP
>>
>> FreeSWITCH Version 1.0.head (git-fccbba5 2011-05-18 19-00-42 -0400)
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From nyamul at gmail.com  Mon Jun 20 21:14:20 2011
From: nyamul at gmail.com (Nyamul Hassan)
Date: Mon, 20 Jun 2011 23:14:20 +0600
Subject: [Freeswitch-users] VoIP IP DSLAMs
In-Reply-To: <BANLkTimJWo21Pi4301-BCCDZk3AuFjzjKA@mail.gmail.com>
References: <0013F8A2-8D82-470C-B8D8-45C22BDB3A00@ipeva.fr>
	<84b3f6b0-9cff-40cc-8a32-f1a2da1458ab@zimbra1.crocker.com>
	<BANLkTimJWo21Pi4301-BCCDZk3AuFjzjKA@mail.gmail.com>
Message-ID: <BANLkTin13NAcM8Uaf1_OftKfyafRJc_Bqg@mail.gmail.com>

On Mon, Jun 20, 2011 at 22:21, Anton VG <anton.vazir at gmail.com> wrote:

> zyxel have what you ask. Huawei have better pricing (~20$ per port)
> but huawei wants to sell huge amounts and quality is somehow bad.
>
> Zyxel is quite good (did not try ZyXEL Voip features, only ADSL) but
> ADSL is good and reliable, but more expensive per port. There are
> options for 12/24/48 and modular 300+ ports
>
> 2011/6/20 Matthew S. Crocker <matthew at corp.crocker.com>:
> >
> > Zhone MALC-XP are nowhere near $11k.    They are < $100/port
> >
> > ________________________________
> >
> > From: "David Ponzone" <david.ponzone at ipeva.fr>
> > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> > Sent: Friday, June 17, 2011 5:29:48 AM
> > Subject: Re: [Freeswitch-users] VoIP IP DSLAMs
> >
> > Hmmm unlikely I think.
> > The Zhone box for instance seems to be around 11,000$.
> > David Ponzone  Direction Technique
> > email: david.ponzone at ipeva.fr
> > tel:      01 74 03 18 97
> > gsm:   06 66 98 76 34
> > Service Client IPeva
> > tel:      0811 46 26 26
> > www.ipeva.fr  -   www.ipeva-studio.com
> > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> > l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion
> > non autoris?e est interdite. Tout message ?lectronique est susceptible
> > d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
> s'il
> > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> >
> >
> >
> > Le 17/06/2011 ? 10:25, Nandy Dagondon a ?crit :
> >
> > yes, it doesn't add up because the application is in a rural area. the
> > Phybridge is a good solution except for the price. thanks David. however,
> > the MSAN, perfectly describes what i'm looking - hopefully the investment
> is
> > lower.
> > thanks to all who contributed.
> >
> > On Fri, Jun 17, 2011 at 3:37 PM, David Ponzone <david.ponzone at ipeva.fr>
> > wrote:
> >>
> >> Nandy,
> >> something does not add up.
> >> When you deploy DSLAMs for public lines, it's not for 24/48 ports, but
> >> rather for 20 000 to 30 000 ports.
> >> Perhaps there is a confusion about what is a public line.
> >> A public line is the 200 meters-8km pair of copper going from the local
> >> (legacy) telco facility to your house.
> >> You don't install your own DSLAMs for 24/48 ports, or perhaps you are in
> a
> >> very specific situation.
> >> I really doubt you will find a cost effective DSLAM for so few ports.
> >> Plus, you realize you will need to install on DSLAM in all the telco
> >> facilities required to get the coverage you need
> >> There are small DSLAMs on the market, but they are targeted for private
> >> lines (hospitality, health care, ...) and I really don't know if they
> can
> >> work on public lines.
> >> In case you actually need a small DSLAM just to run ADSL over a private
> >> pair of copper, you should have a look at:
> >> http://www.phybridge.com/uniphyer-ip-phones.aspx
> >> This thing does not call itself a DSLAM but it is one.
> >> They market that as a "switch" that enables LAN on your copper wires.
> >> It's a small lie for data usage as you only get 25Mbps/1.4Mbps (ATM),
> but
> >> for voice, that's enough.
> >> Basically, you put the "switch"-DSLAM in the network cabinet where the
> >> copper wires go, you connect the wires to it, you connect the ethernet
> port
> >> to your LAN (so your FS). In each room/office, you plug an adapter
> (provided
> >> also by Phybridge) which is basically a small ADSL modem (powered from
> the
> >> switch through the wire), from which you get an ethernet port, with POE
> on
> >> it!
> >> I like the solution on the paper, but never used. Because of the cost:
> >> a 24 ports switch is 3500$
> >> one adapter is 122$
> >> so a full 24 ports solution would be 6500$, which is not very
> competitive
> >> compared to recabling the place.
> >> But your mileage may vary.
> >> David Ponzone  Direction Technique
> >> email: david.ponzone at ipeva.fr
> >> tel:      01 74 03 18 97
> >> gsm:   06 66 98 76 34
> >> Service Client IPeva
> >> tel:      0811 46 26 26
> >> www.ipeva.fr  -   www.ipeva-studio.com
> >> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> >> l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion
> >> non autoris?e est interdite. Tout message ?lectronique est susceptible
> >> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
> s'il
> >> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> >> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> >>
> >>
> >>
> >> Le 17/06/2011 ? 08:28, Nandy Dagondon a ?crit :
> >>
> >> yes, without ATM layer and it's for public lines.
> >>
> >>
> >>
> >> On Fri, Jun 17, 2011 at 2:19 PM, David Ponzone <david.ponzone at ipeva.fr>
> >> wrote:
> >>>
> >>> Which density are you looking for ?
> >>> Is it for use in a private building or on public lines ?
> >>> David Ponzone  Direction Technique
> >>> email: david.ponzone at ipeva.fr
> >>> tel:      01 74 03 18 97
> >>> gsm:   06 66 98 76 34
> >>> Service Client IPeva
> >>> tel:      0811 46 26 26
> >>> www.ipeva.fr  -   www.ipeva-studio.com
> >>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> >>> l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion
> >>> non autoris?e est interdite. Tout message ?lectronique est susceptible
> >>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
> s'il
> >>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de
> ce
> >>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> >>>
> >>>
> >>>
> >>> Le 17/06/2011 ? 08:07, Nandy Dagondon a ?crit :
> >>>
> >>> hello guys!
> >>>
> >>> i'm trying to setup a small exchange w/ Internet service - of course,
> >>> using FS as the exchange. DSLAMs connect to POTS via 2-wire splitters
> and
> >>> that would require using FXS gateway (as what Rob mentioned).
> >>>
> >>> i'm trying to find out if there's a way we can omit the FXS - so it's
> >>> FS-to-DSLAM direct via IP.  i'm not sure if this setup is now possible
> on
> >>> newer IP DSLAMs.
> >>>
> >>>
> >>> On Fri, Jun 17, 2011 at 10:45 AM, jay binks <jaybinks at gmail.com>
> wrote:
> >>>>
> >>>> this is MY take on what Nandy is after.
> >>>> A Dslam provides DSL on certain frequencies of the line..
> >>>> ( some of ) the other frequencies are used for voice.
> >>>> my understanding is that sometimes this is split off to another device
> >>>> to provide the voice,
> >>>> but in this case Nandy is after a DSlam that will do the DSL part AND
> >>>> the voice frequencies ( Voice signalling / audio by SIP / RTP )
> >>>> I guess this is a logical question and would be quite interesting to
> see
> >>>> if there are such devices..
> >>>> sorry I dont know of any :(
> >>>> Jay
> >>>>
> >>>> On Fri, Jun 17, 2011 at 11:31 AM, Rob Hutton <justlikeef at gmail.com>
> >>>> wrote:
> >>>>>
> >>>>> Can you explain further what you are trying to do?  A DSLAM is a
> device
> >>>>> that provides DSL transport, which is independent of what you run
> across it.
> >>>>>  So normally, you have
> >>>>>
> >>>>> |Server | ----|network|----|DSLAM| --------[dsl line] -------[modem
> or
> >>>>> modem/router combo]-----[network]---[Client]
> >>>>>
> >>>>> What the Server is (Could be Freeswitch) and what the Client is
> (could
> >>>>> be a VOIP phone) are independent of the transport???  Onewire, Cisco,
> and
> >>>>> others make some DSL Modem/Router combos with integrated Voip to FXS
> ports
> >>>>> for the CPE end.
> >>>>>
> >>>>> If you find a DSLAM running one of the OSs that Freeswitch supports,
> >>>>> you could compile Freeswitch for it, but I haven't seen one with
> enough CPU
> >>>>> tp handle much besides the traffic.
> >>>>>
> >>>>> On Thursday 16 June 2011 19:03:50 Nandy Dagondon wrote:
> >>>>> > hello everybody,
> >>>>> >
> >>>>> > i'm looking for small (24/48 ports)  IP DSLAMs that inter-connects
> >>>>> > directly
> >>>>> > to FreeSwitch via IP.  is this already available on the market? or
> do
> >>>>> > we
> >>>>> > still have to connect FreeSwitch via POTS splitters and FXS
> gateways?
> >>>>> > i
> >>>>> > appreciate if you can mention some brands/models.
> >>>>> >
> >>>>> > tks,
> >>>>> > nandy
> >>>>>
> >>>>
> >>>>
> >>>> --
> >>>> Sincerely
> >>>>
> >>>> Jay
>

Very informative discussion.  :)

Can someone also share their experience on Fibre Optic based vendors /
solutions?

Regards
HASSAN
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From gmaruzz at gmail.com  Mon Jun 20 21:31:08 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 20 Jun 2011 19:31:08 +0200
Subject: [Freeswitch-users] Skypopen,
 Skype 2.2.0.35 Beta for Linux and Skype Connect
In-Reply-To: <BANLkTimA25tKy+CCgJC6bpEYzw+Pbzs=hQ@mail.gmail.com>
References: <BANLkTim+WjTGiTFNXBdTLqSZ_0t-DVhhmA@mail.gmail.com>
	<BANLkTi=ru21gP4nLpSKSLqMzwBqzwaWHHQ@mail.gmail.com>
	<BANLkTimFPO4JuO0byT4i45C89vG4Teb7GA@mail.gmail.com>
	<BANLkTimA25tKy+CCgJC6bpEYzw+Pbzs=hQ@mail.gmail.com>
Message-ID: <BANLkTi=AcsyA6rsO6zgLN+E9Ck-P0vPpZQ@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Skypopen#SHORT_BLUEPRINT:_STEPS_NEEDED_TO_USE_SKYPOPEN
http://wiki.freeswitch.org/wiki/Skypopen#Build_procedure_.28all_Linuxes.29_for_FS_and_mod_skypopen
http://wiki.freeswitch.org/wiki/Skypopen#How_to_build_skypopen.ko_OSS_sound_driver_.28required.29
http://wiki.freeswitch.org/wiki/Skypopen#Interactive_INSTALLER_and_CONFIGURATOR
http://wiki.freeswitch.org/wiki/Skypopen#How_to_start_the_Skype_clients.2C_then_start_FreeSWITCH_and_load_mod_skypopen_on_Linux



On Mon, Jun 20, 2011 at 6:51 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> All relevant info are in the wiki page ;).
>
> Particularly, I would counseil to use the interactive installer, that
> does it all automatically for you. Included downloading, configuring
> and installing the correct skype client.
>
> -giovanni
>
>
> On 6/20/11, Steven Ayre <steveayre at gmail.com> wrote:
> > I don't think that version's supported...
> >
> > http://wiki.freeswitch.org/wiki/Mod_skypopen*:
> > "Don't* use the *beta* Skype client or more recent "stable" (2.0.0.72
> static
> > for OSS is the one you want to use)."
> >
> > Giovanni is the module maintainer so can probably chip in to clarify if
> > that's still the case or whether that's or information.
> >
> > -Steve
> >
> >
> >
> > On 19 June 2011 12:34, Skyp Tester <skyptester at gmail.com> wrote:
> >
> >> Hi
> >>
> >>
> >>
> >> There is problem with Skype Linux client 2.0.0.72 , I cant call to Skype
> >> Connect client, error ?FAILUREREASON 2? = ?User or phone number does not
> >> exist?.
> >>
> >> http://www.skype.com/intl/en-us/business/skype-connect/
> >>
> >>
> >>
> >> Installed skypopen with Skype client 2.0.0.72 .
> >>
> >> Upgraded to 2.2.0.35:
> >>
> >> 1)      install libstdc++-4.3.2-7.i386.rpm
> >>
> >> 2)      copy over /usr/bin/skype and /usr/share/skype/lang with new
> >> version
> >>
> >>
> >>
> >> Now I can call to Skype Connect client but there are some problems
> >>
> >> 1)      audio is one way, Skype->FS
> >>
> >> 2)      after first call there is error ?NO SPACE WRITE: 640? and no
> more
> >> Skype calls, ?ERROR 68? & ?If I don't connect immediately, please give
> the
> >> Skype client authorization to be connected by Skypopen  (and to not ask
> >> you again)?
> >>
> >>
> >>
> >> Can we hope support to Skype Linux 2.2.0.35 or there is some big
> technical
> >> obstacle?
> >>
> >>
> >>
> >>
> >>
> >> Centos 5.6 x86_64
> >>
> >> 2.6.18-238.12.1.el5 #1 SMP
> >>
> >> FreeSWITCH Version 1.0.head (git-fccbba5 2011-05-18 19-00-42 -0400)
> >>
> >>
> >> _______________________________________________
> >> Join us at ClueCon 2011, Aug 9-11, Chicago
> >> http://www.cluecon.com 877-7-4ACLUE
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
>
> --
> Sent from my mobile device
>
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From msc at freeswitch.org  Mon Jun 20 21:32:42 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 20 Jun 2011 10:32:42 -0700
Subject: [Freeswitch-users] bind_digit_action delay
In-Reply-To: <BANLkTinFcx7Ud=ajEwwffHkc61v8WnTR6w@mail.gmail.com>
References: <BANLkTinFcx7Ud=ajEwwffHkc61v8WnTR6w@mail.gmail.com>
Message-ID: <BANLkTim4N4Qx+XzWhY=ebaLJmshSXwgapg@mail.gmail.com>

Could you please pastebin your dialplan where you set the target actions for
these digits?
-MC

On Mon, Jun 20, 2011 at 4:37 AM, Javier Gallart <jgallartm at gmail.com> wrote:

> Hello list,
>
> I'm using to bind_digit_action application to launch an event when a
> particular sequence of keys (##) is pressed. It works fine, the only problem
> is that the application is launched 5 seconds after the last key is pressed.
> I've tried with a simple playback application and it's just the same:
> 2011-06-20 07:22:55.641019 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF #:1560
> 2011-06-20 07:22:56.121025 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF #:1480
> 2011-06-20 07:22:56.121025 [DEBUG] mod_dptools.c:151 sofia/internal/
> 34917019888 at 79.170.64.151 Digit match binding
> [exec:playback][/var/local/sounds/common/en/last_time]
> (...)
> 2011-06-20 07:22:56.441011 [DEBUG] switch_ivr.c:567 sofia/internal/
> 34661574758 at 79.170.68.169:5060 Command Execute
> playback(local_stream://moh)
> EXECUTE sofia/internal/34661574758 at 79.170.68.169:5060playback(local_stream://moh)
> 2011-06-20 07:22:56.441011 [WARNING] mod_local_stream.c:393 Unknown source
> moh, trying 'default'
> 2011-06-20 07:22:56.441011 [ERR] mod_local_stream.c:402 Unknown source
> default
> (...)
> 2011-06-20 07:23:01.281067 [DEBUG] switch_ivr.c:567 sofia/internal/
> 34917019888 at 79.170.64.151 Command Execute
> playback(/var/local/sounds/common/en/last_time)
> EXECUTE sofia/internal/34917019888 at 79.170.64.151playback(/var/local/sounds/common/en/last_time)
>
> My first question is why does fs tries to play moh? And the second one is
> if there is a way to shorten the lapse between the last key and the
> execution of the application. I've played a bit with the
> bind_digit_input_timeout variable that I've seen in the code, with no
> success. Does anyone have faces the same issue?
> I'm using version FreeSWITCH Version 1.0.head (git-7768808 2011-06-18
> 11-52-37 -0500)
>
> Thanks in advance
>
> Javi
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Jun 20 23:50:55 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 20 Jun 2011 12:50:55 -0700
Subject: [Freeswitch-users] Have someone zrtp with correct sas ?
In-Reply-To: <BAY164-ds10EADF64CEECCFCDCF0F29C16E0@phx.gbl>
References: <BAY164-ds10EADF64CEECCFCDCF0F29C16E0@phx.gbl>
Message-ID: <BANLkTintbY_pUpgEZQvyqW1o=ByH3x8Fdw@mail.gmail.com>

Is media flowing through FreeSWITCH? If so I think that FS needs to be
configured as a "trusted man in the middle." The other option would be to
bypass media and see if that works. I would start with the bypass media
since that is the low-hanging fruit.

-MC

On Sun, Jun 19, 2011 at 7:24 PM, J.B. BERLIN <endlesspixel at hotmail.com>wrote:

>   Dear users,
>
> I try run FS last git build from today 19. June 2011 with ZRTP.
>
> Zfone show me green but the sas are everytime different!
>
> Tested with 2x Notebooks on Fedora 14 Twinkle(via yum) & with two desktops
> with Win7 x64 / Zfone 0.92.build.219 and X-Lite on both sides.
>
> FS is build with libzrtp-0.81.514
>
> zrtp.lua is enabled !
>
> global_setvar zrtp_secure_media=true" is set
>
> Or have someone a real good step by step guide ?
>
> Many thanks and best regards
> Jacek
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From endlesspixel at hotmail.com  Tue Jun 21 00:36:18 2011
From: endlesspixel at hotmail.com (J.B. BERLIN)
Date: Mon, 20 Jun 2011 22:36:18 +0200
Subject: [Freeswitch-users] Have someone zrtp with correct sas ?
In-Reply-To: <BANLkTintbY_pUpgEZQvyqW1o=ByH3x8Fdw@mail.gmail.com>
References: <BAY164-ds10EADF64CEECCFCDCF0F29C16E0@phx.gbl>
	<BANLkTintbY_pUpgEZQvyqW1o=ByH3x8Fdw@mail.gmail.com>
Message-ID: <BAY164-ds8731F6DE9ECCDAD9DAFC6C16E0@phx.gbl>

Hi Michael,

thanks for reply where the difference in the configuration ?

How is the right configuration for bypass and what for MiTM ?

I build FS like so:

sudo git clone git://git.freeswitch.org/freeswitch.git
cd /usr/src/freeswitch
sh build/buildzrtp.sh
sudo ./bootstrap.sh
sudo ./configure -C --prefix=/opt/freeswitch --enable-zrtp --enable-64
sudo make all install sounds-install moh-install

Than put : global_setvar zrtp_secure_media=true in the conf

and following in my dailplan:

<!-- install zrtp_agent.lua into scripts (ZRTP == 9787) -->
      <extension name="zrtp_enrollement">
        <condition field="destination_number" expression="^9787$">
 	<action application="answer"/>
 	<action application="lua" data="zrtp_agent.lua"/>
        </condition>
      </extension>I can call 9787 and it?s say me that the call is secure but if I callanother user i become different SAS...?Kind regardsJacek
From: Michael Collins 
Sent: Monday, June 20, 2011 9:50 PM
To: FreeSWITCH Users Help 
Subject: Re: [Freeswitch-users] Have someone zrtp with correct sas ?

Is media flowing through FreeSWITCH? If so I think that FS needs to be configured as a "trusted man in the middle." The other option would be to bypass media and see if that works. I would start with the bypass media since that is the low-hanging fruit. 

-MC


On Sun, Jun 19, 2011 at 7:24 PM, J.B. BERLIN <endlesspixel at hotmail.com> wrote:

  Dear users,

  I try run FS last git build from today 19. June 2011 with ZRTP.

  Zfone show me green but the sas are everytime different!

  Tested with 2x Notebooks on Fedora 14 Twinkle(via yum) & with two desktops with Win7 x64 / Zfone 0.92.build.219 and X-Lite on both sides.

  FS is build with libzrtp-0.81.514

  zrtp.lua is enabled !

  global_setvar zrtp_secure_media=true" is set

  Or have someone a real good step by step guide ?

  Many thanks and best regards
  Jacek

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

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





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

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From brian at freeswitch.org  Tue Jun 21 00:40:54 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 20 Jun 2011 15:40:54 -0500
Subject: [Freeswitch-users] Have someone zrtp with correct sas ?
In-Reply-To: <BAY164-ds8731F6DE9ECCDAD9DAFC6C16E0@phx.gbl>
References: <BAY164-ds10EADF64CEECCFCDCF0F29C16E0@phx.gbl>
	<BANLkTintbY_pUpgEZQvyqW1o=ByH3x8Fdw@mail.gmail.com>
	<BAY164-ds8731F6DE9ECCDAD9DAFC6C16E0@phx.gbl>
Message-ID: <45A8C155-CAC4-4C87-BA8D-768559BCB9F6@freeswitch.org>

Its called MiTM... and it will switch in some cases depending on who wins. :P

/b

On Jun 20, 2011, at 3:36 PM, J.B. BERLIN wrote:

> Hi Michael,
> 
> thanks for reply where the difference in the configuration ?
> 
> How is the right configuration for bypass and what for MiTM ?
> 
> I build FS like so:



From mitch.johnson7 at gmail.com  Tue Jun 21 05:07:12 2011
From: mitch.johnson7 at gmail.com (Mitch Johnson)
Date: Mon, 20 Jun 2011 21:07:12 -0400
Subject: [Freeswitch-users] FreeSWITCH Cookbook Recipe List
In-Reply-To: <mailman.10883.1308586936.24555.freeswitch-users@lists.freeswitch.org>
References: <mailman.10883.1308586936.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <16591747-C9A1-4B88-AC4D-8C986EE5A961@gmail.com>

I would love to supply examples and screenshots, but to be honest it would feel like stealing from the work of others such as Brian and the other Mitch (Capper).  They have both helped me to be successful in this area.  Brian was a huge help on the Cudatel side and Mitch was incredible on the Freeswitch side.  I did read Brian's response and he is right, there are examples available, but having a complete explanation on using one channel for encrypted and the other as unencrypted.  Or having both channels encrypted.  An explanation on the process of how to build a cert server and then import certificates into a mobile device.  

Once again, I have done these things, but I am hardly the right person to be able to explain the process in an effective manner.  The other Mitch (Capper) explained it brilliantly to me and Brian West had to reiterate everything that Mitch explained.

If there is anything I can do to make FreeSWITCH the premier TLS/SRTP solution I would love to help.  I think all I could offer is the way I eventually connected FreeSWITCH to the Cisco CallManager platform.  That part I think I can provide screenshots.  

Thanks for everyone's help, especially Mitch and Brian.

Mitch (Not Capper)

From bcxml at hotmail.com  Tue Jun 21 06:29:43 2011
From: bcxml at hotmail.com (Brian Campbell)
Date: Mon, 20 Jun 2011 22:29:43 -0400
Subject: [Freeswitch-users] Intermittent Audio Problem - Incomming Calls
Message-ID: <SNT135-w24A462340F95032961EE0FB7510@phx.gbl>


I have run into a very strange issue with periodically losing Audio on Incomming calls
 
I am receiving incomming calls to FreeSwitch via a DID from an ITSP. Once FreeSwitch receives the call, I am adding a header to pass the UUID (which I use later to inject some custom values into CDRs) and then bridge to an IVR application running on Microsoft Speech Server 2007 which is on a separate box, using the following dial plan
 
<extension name="Inbound_FromSoTel">
  <condition field="destination_number" expression="^1(\d+)$">
    <action application="set" data="sip_h_X-FS_UUID=${uuid}" /> 
    <action application="bridge" data="sofia/external/$1 at 173.14.17.213:5060;transport=tcp" /> 
  </condition>
</extension>
 
Some calls are performing properly, but in some of the calls (about 30%) the IVR application doesn't seem to hear anything the caller is saying
 
I have eliminated any Firewalls for testing, so I go straight from the ITSP to FreeSwitch and then to Speech Server. Thee is no NAT in play.
 
There doesn't seem to be any rhyme or reason as to when a call is going to work, or when it isn't. I have also seen exactly the same behavior in a completely different environment with a different ITSP and a completely different IVR application.
 
So I figure that the problem is somewhere between FreeSwitch and Speech Server, but I don't see anything wrong. 
 
Using WireShark I can see the SIP signalling creating/tearing down the call and I see RTP traffic flowing both ways between FreeSwitch and Speech Server
 
The version of FreeSwitch that I am running is 'FreeSWITCH Version 1.0.head (git-38f06a3 2011-05-19 15-39-43 -0500)'
 
I have pasted 2 FreeSwitch traces into pastebin. The first one is of a working call, the second one is of a call where the IVR application doesn't seem to hear anything the caller is saying
 
Can anyone advise on what the issue might be. I am completely out of ideas.
 
Thanks
 
Brian Campbell

  		 	   		  
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From david.ponzone at ipeva.fr  Tue Jun 21 10:43:44 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 21 Jun 2011 08:43:44 +0200
Subject: [Freeswitch-users] Intermittent Audio Problem - Incomming Calls
In-Reply-To: <SNT135-w24A462340F95032961EE0FB7510@phx.gbl>
References: <SNT135-w24A462340F95032961EE0FB7510@phx.gbl>
Message-ID: <EEBB17CE-15B0-4D72-A849-61505092D995@ipeva.fr>

Brian,

twice you say "the IVR doesn't seem to hear anything the caller is saying".
I suppose that means it is an assumption because it does not react to DTMFs ?
Do you have the possibility in this IVR to record the call ?
This way, you would be sure.

Also you say that Wireshark tells you the RTP traffic is fine.
Even when the call is not working ?
Do you notice anything different in this RTP ?
You may also take a RTP trace between FS and IVR of a non-working call, analyse the RTP flows and save the payload as raw to a file.
You should be able to listen to the audio with the right tool.

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

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

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




Le 21/06/2011 ? 04:29, Brian Campbell a ?crit :

> I have run into a very strange issue with periodically losing Audio on Incomming calls
>  
> I am receiving incomming calls to FreeSwitch via a DID from an ITSP. Once FreeSwitch receives the call, I am adding a header to pass the UUID (which I use later to inject some custom values into CDRs) and then bridge to an IVR application running on Microsoft Speech Server 2007 which is on a separate box, using the following dial plan
>  
> <extension name="Inbound_FromSoTel">
>   <condition field="destination_number" expression="^1(\d+)$">
>     <action application="set" data="sip_h_X-FS_UUID=${uuid}" /> 
>     <action application="bridge" data="sofia/external/$1 at 173.14.17.213:5060;transport=tcp" /> 
>   </condition>
> </extension>
>  
> Some calls are performing properly, but in some of the calls (about 30%) the IVR application doesn't seem to hear anything the caller is saying
>  
> I have eliminated any Firewalls for testing, so I go straight from the ITSP to FreeSwitch and then to Speech Server. Thee is no NAT in play.
>  
> There doesn't seem to be any rhyme or reason as to when a call is going to work, or when it isn't. I have also seen exactly the same behavior in a completely different environment with a different ITSP and a completely different IVR application.
>  
> So I figure that the problem is somewhere between FreeSwitch and Speech Server, but I don't see anything wrong. 
>  
> Using WireShark I can see the SIP signalling creating/tearing down the call and I see RTP traffic flowing both ways between FreeSwitch and Speech Server
>  
> The version of FreeSwitch that I am running is 'FreeSWITCH Version 1.0.head (git-38f06a3 2011-05-19 15-39-43 -0500)'
>  
> I have pasted 2 FreeSwitch traces into pastebin. The first one is of a working call, the second one is of a call where the IVR application doesn't seem to hear anything the caller is saying
>  
> Can anyone advise on what the issue might be. I am completely out of ideas.
>  
> Thanks
>  
> Brian Campbell
> 
>  
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jgallartm at gmail.com  Tue Jun 21 14:18:29 2011
From: jgallartm at gmail.com (Javier Gallart)
Date: Tue, 21 Jun 2011 12:18:29 +0200
Subject: [Freeswitch-users] bind_digit_action delay
Message-ID: <BANLkTikty5EGSxWEcPPBHqAb=pkSKbW-Kg@mail.gmail.com>

Thanks Michael

the session has been answered by a lua script and transferred back to the
dialplan to be bridged. Once in the dialplan this is the extension executed.
The CUSTOM schedule event is sent 5 seconds after the ## is pressed.

<extension name="bridge_session" continue="false">
        <condition field="${call_type}" expression="prepaid">
            <action application="set" data="continue_on_fail=1"/>
            <action application="export" data="execute_on_answer=event
Event-Subclass=schedule,Event-Name=CUSTOM"/>
            (...here we assign some custom variables...)
            <action application="export"
data="bridge_pre_execute_aleg_app=bind_digit_action"/>
            <action application="export"
data="bridge_pre_execute_aleg_data=test1,##,exec:event,Event-Subclass=reoriginate,Event-Name=CUSTOM"/>
            <action application="digit_action_set_realm" data="test1"/>
            <action application="bridge"
data="sofia/internal/${dest_number}@${gateway}"/>
        </condition>
</extension>

Regards

Javi



---------- Forwarded message ----------
> From: Michael Collins <msc at freeswitch.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Mon, 20 Jun 2011 10:32:42 -0700
> Subject: Re: [Freeswitch-users] bind_digit_action delay
> Could you please pastebin your dialplan where you set the target actions
> for these digits?
> -MC
>
> On Mon, Jun 20, 2011 at 4:37 AM, Javier Gallart <jgallartm at gmail.com>wrote:
>
>> Hello list,
>>
>> I'm using to bind_digit_action application to launch an event when a
>> particular sequence of keys (##) is pressed. It works fine, the only problem
>> is that the application is launched 5 seconds after the last key is pressed.
>> I've tried with a simple playback application and it's just the same:
>> 2011-06-20 07:22:55.641019 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF #:1560
>> 2011-06-20 07:22:56.121025 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF #:1480
>> 2011-06-20 07:22:56.121025 [DEBUG] mod_dptools.c:151 sofia/internal/
>> 34917019888 at 79.170.64.151 Digit match binding
>> [exec:playback][/var/local/sounds/common/en/last_time]
>> (...)
>> 2011-06-20 07:22:56.441011 [DEBUG] switch_ivr.c:567 sofia/internal/
>> 34661574758 at 79.170.68.169:5060 Command Execute
>> playback(local_stream://moh)
>> EXECUTE sofia/internal/34661574758 at 79.170.68.169:5060playback(local_stream://moh)
>> 2011-06-20 07:22:56.441011 [WARNING] mod_local_stream.c:393 Unknown source
>> moh, trying 'default'
>> 2011-06-20 07:22:56.441011 [ERR] mod_local_stream.c:402 Unknown source
>> default
>> (...)
>> 2011-06-20 07:23:01.281067 [DEBUG] switch_ivr.c:567 sofia/internal/
>> 34917019888 at 79.170.64.151 Command Execute
>> playback(/var/local/sounds/common/en/last_time)
>> EXECUTE sofia/internal/34917019888 at 79.170.64.151playback(/var/local/sounds/common/en/last_time)
>>
>> My first question is why does fs tries to play moh? And the second one is
>> if there is a way to shorten the lapse between the last key and the
>> execution of the application. I've played a bit with the
>> bind_digit_input_timeout variable that I've seen in the code, with no
>> success. Does anyone have faces the same issue?
>> I'm using version FreeSWITCH Version 1.0.head (git-7768808 2011-06-18
>> 11-52-37 -0500)
>>
>> Thanks in advance
>>
>> Javi
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
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From marketing at cluecon.com  Tue Jun 21 16:39:42 2011
From: marketing at cluecon.com (marketing at cluecon.com)
Date: Tue, 21 Jun 2011 12:39:42 +0000
Subject: [Freeswitch-users] ClueCon 2011 Special Guest: PGP Inventor Philip
	Zimmermann
Message-ID: <00000130b236a04b-f80474f5-f26e-4ddb-820c-ddadbecd3c08-000000@email.amazonses.com>

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From bcxml at hotmail.com  Tue Jun 21 17:13:19 2011
From: bcxml at hotmail.com (Brian Campbell)
Date: Tue, 21 Jun 2011 09:13:19 -0400
Subject: [Freeswitch-users] Intermittent Audio Problem - Incomming Calls
In-Reply-To: <EEBB17CE-15B0-4D72-A849-61505092D995@ipeva.fr>
References: <SNT135-w24A462340F95032961EE0FB7510@phx.gbl>,
	<EEBB17CE-15B0-4D72-A849-61505092D995@ipeva.fr>
Message-ID: <SNT135-w49311427DD49F93EF3280DB7510@phx.gbl>


Thanks David

The Speech Server application reacts to speech from the caller validated by a grammar. There is a facility in Speech Server to log the calls, including the audio. On calls that work, I can hear the prompts from the application as well as my responses in the session audio. On calls that don't work, I hear the prompts from the application but none of my responses. After a small period of waiting the Speech Recognizer times out and tells me that it did not hear me say anything. I have tried increasing the timeout interval but the problem remains.
 
So I am not sure where the problem lies, is it something breaking down in Speech Server, or is the audio not making it from FreeSwitch to Speech Server. 
 
One thing for sure is that it doesnt stop working half way through the call. It either recognizes utterances from the caller or it doesn't, which would lead me to think that sometimes the call is not properly setup between Speech Server and FreeSwitch, but as I said I have run out of ideas. 
 
As I mentioned I do see RTP traffic, but I am afraid that I do not posses indepth knowledge about the protocol, so I am not sure how to properly analyse it.
 
Thanks again for your input
 
 
Brian Campbell
 



From: david.ponzone at ipeva.fr
Date: Tue, 21 Jun 2011 08:43:44 +0200
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Intermittent Audio Problem - Incomming Calls


Brian,


twice you say "the IVR doesn't seem to hear anything the caller is saying".
I suppose that means it is an assumption because it does not react to DTMFs ?
Do you have the possibility in this IVR to record the call ?
This way, you would be sure.


Also you say that Wireshark tells you the RTP traffic is fine.
Even when the call is not working ?
Do you notice anything different in this RTP ?
You may also take a RTP trace between FS and IVR of a non-working call, analyse the RTP flows and save the payload as raw to a file.
You should be able to listen to the audio with the right tool.




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


Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr  -   www.ipeva-studio.com



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




Le 21/06/2011 ? 04:29, Brian Campbell a ?crit :


I have run into a very strange issue with periodically losing Audio on Incomming calls
 
I am receiving incomming calls to FreeSwitch via a DID from an ITSP. Once FreeSwitch receives the call, I am adding a header to pass the UUID (which I use later to inject some custom values into CDRs) and then bridge to an IVR application running on Microsoft Speech Server 2007 which is on a separate box, using the following dial plan
 
<extension name="Inbound_FromSoTel">
  <condition field="destination_number" expression="^1(\d+)$">
    <action application="set" data="sip_h_X-FS_UUID=${uuid}" /> 
    <action application="bridge" data="sofia/external/$1 at 173.14.17.213:5060;transport=tcp" /> 
  </condition>
</extension>
 
Some calls are performing properly, but in some of the calls (about 30%) the IVR application doesn't seem to hear anything the caller is saying
 
I have eliminated any Firewalls for testing, so I go straight from the ITSP to FreeSwitch and then to Speech Server. Thee is no NAT in play.
 
There doesn't seem to be any rhyme or reason as to when a call is going to work, or when it isn't. I have also seen exactly the same behavior in a completely different environment with a different ITSP and a completely different IVR application.
 
So I figure that the problem is somewhere between FreeSwitch and Speech Server, but I don't see anything wrong. 
 
Using WireShark I can see the SIP signalling creating/tearing down the call and I see RTP traffic flowing both ways between FreeSwitch and Speech Server
 
The version of FreeSwitch that I am running is 'FreeSWITCH Version 1.0.head (git-38f06a3 2011-05-19 15-39-43 -0500)'
 
I have pasted 2 FreeSwitch traces into pastebin. The first one is of a working call, the second one is of a call where the IVR application doesn't seem to hear anything the caller is saying
 
Can anyone advise on what the issue might be. I am completely out of ideas.
 
Thanks
 
Brian Campbell

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

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


_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 		 	   		  
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From henk at oegema.com  Tue Jun 21 17:48:08 2011
From: henk at oegema.com (Henk Oegema)
Date: Tue, 21 Jun 2011 15:48:08 +0200
Subject: [Freeswitch-users] Call internal, otherwise mobile, not working
Message-ID: <1308664088.9892.86.camel@DELL>

I want to achieve the following: 
When a call comes in, it should ring 30 seconds on extension 1000.
If there's no answer the call should go to my mobile.


............... 
<action application="set" data="continue_on_fail=true"/>
 <action application="bridge"
data="{originate_timeout=30}sofia/internal/1000%$${domain}"/>
 <action application="set" data="hangup_after_bridge=true"/>
 <action application="bridge"
data="sofia/gateway/powervoip/003163086xxxx"/>
 <action application="voicemail" data="default $${domain} 1000"/>
.............


However,  when I answer the incoming call at 1000 and hangup, the call
will still  go to my mobile, in stead of hanging up.
What am I doing wrong?

Rgds.
Henk
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From monemran at gmail.com  Tue Jun 21 17:56:48 2011
From: monemran at gmail.com (Mohammad Emran)
Date: Tue, 21 Jun 2011 19:56:48 +0600
Subject: [Freeswitch-users] Call internal, otherwise mobile, not working
In-Reply-To: <1308664088.9892.86.camel@DELL>
References: <1308664088.9892.86.camel@DELL>
Message-ID: <6F677925-0E82-4A82-9477-7347C8AF25FA@gmail.com>

You can try to set hangup_after_bridge true before bridge.

Sent from my iPad

On 21 Jun 2011, at 19:48, Henk Oegema <henk at oegema.com> wrote:

> I want to achieve the following: 
> When a call comes in, it should ring 30 seconds on extension 1000.
> If there's no answer the call should go to my mobile.
> 
> 
> ............... 
> <action application="set" data="continue_on_fail=true"/>
> <action application="bridge" data="{originate_timeout=30}sofia/internal/1000%$${domain}"/>
> <action application="set" data="hangup_after_bridge=true"/>
> <action application="bridge" data="sofia/gateway/powervoip/003163086xxxx"/>
> <action application="voicemail" data="default $${domain} 1000"/>
> .............
> 
> 
> However,  when I answer the incoming call at 1000 and hangup, the call will still  go to my mobile, in stead of hanging up.
> What am I doing wrong?
> 
> Rgds.
> Henk
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From avi at avimarcus.net  Tue Jun 21 18:06:20 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 21 Jun 2011 17:06:20 +0300
Subject: [Freeswitch-users] Call internal, otherwise mobile, not working
In-Reply-To: <6F677925-0E82-4A82-9477-7347C8AF25FA@gmail.com>
References: <1308664088.9892.86.camel@DELL>
	<6F677925-0E82-4A82-9477-7347C8AF25FA@gmail.com>
Message-ID: <BANLkTimwx2r2so8gOnxcwXy78dJEm_LtTA@mail.gmail.com>

Not just "try" :)
You want it to hangup_after_bridge, so you need to tell it that before the
first bridge.

 -Avi Marcus


On Tue, Jun 21, 2011 at 4:56 PM, Mohammad Emran <monemran at gmail.com> wrote:

> You can try to set hangup_after_bridge true before bridge.
>
> Sent from my iPad
>
> On 21 Jun 2011, at 19:48, Henk Oegema <henk at oegema.com> wrote:
>
> > I want to achieve the following:
> > When a call comes in, it should ring 30 seconds on extension 1000.
> > If there's no answer the call should go to my mobile.
> >
> >
> > ...............
> > <action application="set" data="continue_on_fail=true"/>
> > <action application="bridge"
> data="{originate_timeout=30}sofia/internal/1000%$${domain}"/>
> > <action application="set" data="hangup_after_bridge=true"/>
> > <action application="bridge"
> data="sofia/gateway/powervoip/003163086xxxx"/>
> > <action application="voicemail" data="default $${domain} 1000"/>
> > .............
> >
> >
> > However,  when I answer the incoming call at 1000 and hangup, the call
> will still  go to my mobile, in stead of hanging up.
> > What am I doing wrong?
> >
> > Rgds.
> > Henk
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From a.luppi at seletech.com  Tue Jun 21 18:09:32 2011
From: a.luppi at seletech.com (Alessandro)
Date: Tue, 21 Jun 2011 16:09:32 +0200
Subject: [Freeswitch-users] Modified Follow me
Message-ID: <4E00A61C.1060503@seletech.com>

Hi,

i'd like to implement this particular behaviour. If I call an extension 
that isn't registered to FS, FS re-route the call to another 
destination. If also the second destination in unreachable the call end. 
I first used the follow me (created with fusionpbx) to implement this 
feature, but I have the following problem: if the first extension is 
reachable and the user hangup before answering (hangup in ringing), the 
follow me procedure starts to contact the second extension in the list. 
I want that FS start to contact the second phone only if the first 
extension isn't registered to FS.
The second problem is that with follow-me FS answer with 200 OK at the 
invite of the caller bypassing the ringing status, than the procedure of 
follow-me is transparent to the caller.
I'd like that the caller stay in ringing status until one of the 
destination extension answer at the call or the call is closed by FS.
Any suggestion?

Regards

Alessandro Luppi

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



From henk at oegema.com  Tue Jun 21 18:25:07 2011
From: henk at oegema.com (Henk Oegema)
Date: Tue, 21 Jun 2011 16:25:07 +0200
Subject: [Freeswitch-users] Call internal, otherwise mobile, not working
In-Reply-To: <BANLkTimwx2r2so8gOnxcwXy78dJEm_LtTA@mail.gmail.com>
References: <1308664088.9892.86.camel@DELL>
	<6F677925-0E82-4A82-9477-7347C8AF25FA@gmail.com>
	<BANLkTimwx2r2so8gOnxcwXy78dJEm_LtTA@mail.gmail.com>
Message-ID: <1308666307.10396.1.camel@DELL>

You are right of course.

On Tue, 2011-06-21 at 17:06 +0300, Avi Marcus wrote:
> Not just "try" :)
> 
> You want it to hangup_after_bridge, so you need to tell it that before
> the first bridge.
> 
> 
> -Avi Marcus
> 
> 
> 
> 
> 
> 
> 
> On Tue, Jun 21, 2011 at 4:56 PM, Mohammad Emran <monemran at gmail.com>
> wrote:
> 
>         You can try to set hangup_after_bridge true before bridge.
>         
>         Sent from my iPad
>         
>         
>         
>         On 21 Jun 2011, at 19:48, Henk Oegema <henk at oegema.com> wrote:
>         
>         > I want to achieve the following:
>         > When a call comes in, it should ring 30 seconds on extension
>         1000.
>         > If there's no answer the call should go to my mobile.
>         >
>         >
>         > ...............
>         > <action application="set" data="continue_on_fail=true"/>
>         > <action application="bridge"
>         data="{originate_timeout=30}sofia/internal/1000%$${domain}"/>
>         > <action application="set" data="hangup_after_bridge=true"/>
>         > <action application="bridge"
>         data="sofia/gateway/powervoip/003163086xxxx"/>
>         > <action application="voicemail" data="default $${domain}
>         1000"/>
>         > .............
>         >
>         >
>         > However,  when I answer the incoming call at 1000 and
>         hangup, the call will still  go to my mobile, in stead of
>         hanging up.
>         > What am I doing wrong?
>         >
>         > Rgds.
>         > Henk
>         
>         
>         > _______________________________________________
>         > Join us at ClueCon 2011, Aug 9-11, Chicago
>         > http://www.cluecon.com 877-7-4ACLUE
>         >
>         > FreeSWITCH-users mailing list
>         > FreeSWITCH-users at lists.freeswitch.org
>         >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         > http://www.freeswitch.org
>         
>         _______________________________________________
>         Join us at ClueCon 2011, Aug 9-11, Chicago
>         http://www.cluecon.com 877-7-4ACLUE
>         
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
> 
> 
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From david.ponzone at ipeva.fr  Tue Jun 21 18:41:14 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 21 Jun 2011 16:41:14 +0200
Subject: [Freeswitch-users] Intermittent Audio Problem - Incomming Calls
In-Reply-To: <SNT135-w49311427DD49F93EF3280DB7510@phx.gbl>
References: <SNT135-w24A462340F95032961EE0FB7510@phx.gbl>,
	<EEBB17CE-15B0-4D72-A849-61505092D995@ipeva.fr>
	<SNT135-w49311427DD49F93EF3280DB7510@phx.gbl>
Message-ID: <E811A8E4-8A8A-4EE3-BE8A-4A26B6E6C759@ipeva.fr>

Wireshark is your friend :)
Open your RTP flow, click on Telephony/RTP/Show All Streams
In the next window, select the stream you want (the one from FS to IVR) and click on Analyse.
And in the final window, click on Save Payload.
Then select the forward stream, and save as raw.

You should be able to import the raw file in any decent audio software (Audacity, etc...) as U-Law (I suppose you use PCMU), 8Khz.

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

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

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




Le 21/06/2011 ? 15:13, Brian Campbell a ?crit :

> Thanks David
> 
> The Speech Server application reacts to speech from the caller validated by a grammar. There is a facility in Speech Server to log the calls, including the audio. On calls that work, I can hear the prompts from the application as well as my responses in the session audio. On calls that don't work, I hear the prompts from the application but none of my responses. After a small period of waiting the Speech Recognizer times out and tells me that it did not hear me say anything. I have tried increasing the timeout interval but the problem remains.
>  
> So I am not sure where the problem lies, is it something breaking down in Speech Server, or is the audio not making it from FreeSwitch to Speech Server. 
>  
> One thing for sure is that it doesnt stop working half way through the call. It either recognizes utterances from the caller or it doesn't, which would lead me to think that sometimes the call is not properly setup between Speech Server and FreeSwitch, but as I said I have run out of ideas. 
>  
> As I mentioned I do see RTP traffic, but I am afraid that I do not posses indepth knowledge about the protocol, so I am not sure how to properly analyse it.
>  
> Thanks again for your input
>  
>  
> Brian Campbell
>  
> From: david.ponzone at ipeva.fr
> Date: Tue, 21 Jun 2011 08:43:44 +0200
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Intermittent Audio Problem - Incomming Calls
> 
> Brian,
> 
> twice you say "the IVR doesn't seem to hear anything the caller is saying".
> I suppose that means it is an assumption because it does not react to DTMFs ?
> Do you have the possibility in this IVR to record the call ?
> This way, you would be sure.
> 
> Also you say that Wireshark tells you the RTP traffic is fine.
> Even when the call is not working ?
> Do you notice anything different in this RTP ?
> You may also take a RTP trace between FS and IVR of a non-working call, analyse the RTP flows and save the payload as raw to a file.
> You should be able to listen to the audio with the right tool.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 21/06/2011 ? 04:29, Brian Campbell a ?crit :
> 
> I have run into a very strange issue with periodically losing Audio on Incomming calls
>  
> I am receiving incomming calls to FreeSwitch via a DID from an ITSP. Once FreeSwitch receives the call, I am adding a header to pass the UUID (which I use later to inject some custom values into CDRs) and then bridge to an IVR application running on Microsoft Speech Server 2007 which is on a separate box, using the following dial plan
>  
> <extension name="Inbound_FromSoTel">
>   <condition field="destination_number" expression="^1(\d+)$">
>     <action application="set" data="sip_h_X-FS_UUID=${uuid}" /> 
>     <action application="bridge" data="sofia/external/$1 at 173.14.17.213:5060;transport=tcp" /> 
>   </condition>
> </extension>
>  
> Some calls are performing properly, but in some of the calls (about 30%) the IVR application doesn't seem to hear anything the caller is saying
>  
> I have eliminated any Firewalls for testing, so I go straight from the ITSP to FreeSwitch and then to Speech Server. Thee is no NAT in play.
>  
> There doesn't seem to be any rhyme or reason as to when a call is going to work, or when it isn't. I have also seen exactly the same behavior in a completely different environment with a different ITSP and a completely different IVR application.
>  
> So I figure that the problem is somewhere between FreeSwitch and Speech Server, but I don't see anything wrong. 
>  
> Using WireShark I can see the SIP signalling creating/tearing down the call and I see RTP traffic flowing both ways between FreeSwitch and Speech Server
>  
> The version of FreeSwitch that I am running is 'FreeSWITCH Version 1.0.head (git-38f06a3 2011-05-19 15-39-43 -0500)'
>  
> I have pasted 2 FreeSwitch traces into pastebin. The first one is of a working call, the second one is of a call where the IVR application doesn't seem to hear anything the caller is saying
>  
> Can anyone advise on what the issue might be. I am completely out of ideas.
>  
> Thanks
>  
> Brian Campbell
> 
>  
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From jgallartm at gmail.com  Tue Jun 21 19:40:43 2011
From: jgallartm at gmail.com (Javier Gallart)
Date: Tue, 21 Jun 2011 17:40:43 +0200
Subject: [Freeswitch-users] bind_digit_action delay
Message-ID: <BANLkTimCkq1fLB3UarQWt-ndv6i1sK3T+A@mail.gmail.com>

Hi

you can ignore this issue, it was a stupid fault in my config. The
/usr/local/freeswitch/sounds dir was empty (we have our custom sounds in a
different path), but at vars.xml the following directives were still
present:
<X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/>
<X-PRE-PROCESS cmd="set"
data="sound_prefix=/usr/local/freeswitch/sounds/en/us/callie"/>

Thanks anyway


Javi

>
> ---------- Forwarded message ----------
> From: Javier Gallart <jgallartm at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Tue, 21 Jun 2011 12:18:29 +0200
> Subject: Re: [Freeswitch-users] bind_digit_action delay
> Thanks Michael
>
> the session has been answered by a lua script and transferred back to the
> dialplan to be bridged. Once in the dialplan this is the extension executed.
> The CUSTOM schedule event is sent 5 seconds after the ## is pressed.
>
> <extension name="bridge_session" continue="false">
>         <condition field="${call_type}" expression="prepaid">
>             <action application="set" data="continue_on_fail=1"/>
>             <action application="export" data="execute_on_answer=event
> Event-Subclass=schedule,Event-Name=CUSTOM"/>
>             (...here we assign some custom variables...)
>             <action application="export"
> data="bridge_pre_execute_aleg_app=bind_digit_action"/>
>             <action application="export"
> data="bridge_pre_execute_aleg_data=test1,##,exec:event,Event-Subclass=reoriginate,Event-Name=CUSTOM"/>
>             <action application="digit_action_set_realm" data="test1"/>
>             <action application="bridge"
> data="sofia/internal/${dest_number}@${gateway}"/>
>         </condition>
> </extension>
>
> Regards
>
> Javi
>
>
>
> ---------- Forwarded message ----------
>> From: Michael Collins <msc at freeswitch.org>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Date: Mon, 20 Jun 2011 10:32:42 -0700
>> Subject: Re: [Freeswitch-users] bind_digit_action delay
>> Could you please pastebin your dialplan where you set the target actions
>> for these digits?
>> -MC
>>
>> On Mon, Jun 20, 2011 at 4:37 AM, Javier Gallart <jgallartm at gmail.com>wrote:
>>
>>> Hello list,
>>>
>>> I'm using to bind_digit_action application to launch an event when a
>>> particular sequence of keys (##) is pressed. It works fine, the only problem
>>> is that the application is launched 5 seconds after the last key is pressed.
>>> I've tried with a simple playback application and it's just the same:
>>> 2011-06-20 07:22:55.641019 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF #:1560
>>> 2011-06-20 07:22:56.121025 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF #:1480
>>> 2011-06-20 07:22:56.121025 [DEBUG] mod_dptools.c:151 sofia/internal/
>>> 34917019888 at 79.170.64.151 Digit match binding
>>> [exec:playback][/var/local/sounds/common/en/last_time]
>>> (...)
>>> 2011-06-20 07:22:56.441011 [DEBUG] switch_ivr.c:567 sofia/internal/
>>> 34661574758 at 79.170.68.169:5060 Command Execute
>>> playback(local_stream://moh)
>>> EXECUTE sofia/internal/34661574758 at 79.170.68.169:5060playback(local_stream://moh)
>>> 2011-06-20 07:22:56.441011 [WARNING] mod_local_stream.c:393 Unknown
>>> source moh, trying 'default'
>>> 2011-06-20 07:22:56.441011 [ERR] mod_local_stream.c:402 Unknown source
>>> default
>>> (...)
>>> 2011-06-20 07:23:01.281067 [DEBUG] switch_ivr.c:567 sofia/internal/
>>> 34917019888 at 79.170.64.151 Command Execute
>>> playback(/var/local/sounds/common/en/last_time)
>>> EXECUTE sofia/internal/34917019888 at 79.170.64.151playback(/var/local/sounds/common/en/last_time)
>>>
>>> My first question is why does fs tries to play moh? And the second one is
>>> if there is a way to shorten the lapse between the last key and the
>>> execution of the application. I've played a bit with the
>>> bind_digit_input_timeout variable that I've seen in the code, with no
>>> success. Does anyone have faces the same issue?
>>> I'm using version FreeSWITCH Version 1.0.head (git-7768808 2011-06-18
>>> 11-52-37 -0500)
>>>
>>> Thanks in advance
>>>
>>> Javi
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From avi at avimarcus.net  Tue Jun 21 19:50:23 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 21 Jun 2011 18:50:23 +0300
Subject: [Freeswitch-users] Fax API
Message-ID: <BANLkTimAB_QwG5BHvTp=gvgpny1=5zWiwg@mail.gmail.com>

I've seen Faxing issues over and over in the channel, so since I don't need
it so much I'd like to outsource it.
I've had a recommendation to use metrofax - they seem to have a nice
website, great pricing, and an API that I may be able to get access to.
Are there any other suggestions for a faxing both inbound and outbound that
I can plug into my GUI for customers?

Thanks!

-Avi Marcus
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From rhuddleston at gmail.com  Tue Jun 21 20:09:30 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Tue, 21 Jun 2011 12:09:30 -0400
Subject: [Freeswitch-users] Fax API
In-Reply-To: <BANLkTimAB_QwG5BHvTp=gvgpny1=5zWiwg@mail.gmail.com>
References: <BANLkTimAB_QwG5BHvTp=gvgpny1=5zWiwg@mail.gmail.com>
Message-ID: <0cdb01cc302d$9aeb8690$d0c293b0$@com>

Tried hylafax?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi
Marcus
Sent: Tuesday, June 21, 2011 11:50 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Fax API

 

I've seen Faxing issues over and over in the channel, so since I don't need
it so much I'd like to outsource it.

I've had a recommendation to use metrofax - they seem to have a nice
website, great pricing, and an API that I may be able to get access to.

Are there any other suggestions for a faxing both inbound and outbound that
I can plug into my GUI for customers?

 

Thanks!




-Avi Marcus

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From avi at avimarcus.net  Tue Jun 21 20:26:21 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 21 Jun 2011 19:26:21 +0300
Subject: [Freeswitch-users] Fax API
In-Reply-To: <0cdb01cc302d$9aeb8690$d0c293b0$@com>
References: <BANLkTimAB_QwG5BHvTp=gvgpny1=5zWiwg@mail.gmail.com>
	<0cdb01cc302d$9aeb8690$d0c293b0$@com>
Message-ID: <BANLkTik=sEwqJRXp0wztmTJ2kJxZPmWmYA@mail.gmail.com>

That's interesting, dedicated faxing software.
Actually though I'm looking for a hosted kind of solution.. just POST the
pdf file and I don't need to worry about any sort of debugging beyond that.

On Tue, Jun 21, 2011 at 7:09 PM, Robert Huddleston <rhuddleston at gmail.com>wrote:

> Tried hylafax?****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus
> *Sent:* Tuesday, June 21, 2011 11:50 AM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] Fax API****
>
> ** **
>
> I've seen Faxing issues over and over in the channel, so since I don't need
> it so much I'd like to outsource it.****
>
> I've had a recommendation to use metrofax - they seem to have a nice
> website, great pricing, and an API that I may be able to get access to.***
> *
>
> Are there any other suggestions for a faxing both inbound and outbound that
> I can plug into my GUI for customers?****
>
> ** **
>
> Thanks!****
>
>
> ****
>
> -Avi Marcus****
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cmcureau at gmail.com  Tue Jun 21 20:36:59 2011
From: cmcureau at gmail.com (Chris Cureau)
Date: Tue, 21 Jun 2011 11:36:59 -0500
Subject: [Freeswitch-users] Fax API
In-Reply-To: <BANLkTik=sEwqJRXp0wztmTJ2kJxZPmWmYA@mail.gmail.com>
References: <BANLkTimAB_QwG5BHvTp=gvgpny1=5zWiwg@mail.gmail.com>
	<0cdb01cc302d$9aeb8690$d0c293b0$@com>
	<BANLkTik=sEwqJRXp0wztmTJ2kJxZPmWmYA@mail.gmail.com>
Message-ID: <-7597423912234016994@unknownmsgid>

Avi,

I am sure that I can provide a solution for you. My company is still in
start-up mode, but I can provide you DIDs for inbound fax and accept PDFs
for delivery. My current architecture does T.38 from the switch to the
customer, so we can work with ATAs as well

Want to discuss? Drop me a line!

--
Chris Cureau
President
eTech Data Solutions, Inc.
(985) 707-6510

On Jun 21, 2011, at 11:30 AM, Avi Marcus <avi at avimarcus.net> wrote:

That's interesting, dedicated faxing software.
Actually though I'm looking for a hosted kind of solution.. just POST the
pdf file and I don't need to worry about any sort of debugging beyond that.

On Tue, Jun 21, 2011 at 7:09 PM, Robert Huddleston <rhuddleston at gmail.com>wrote:

> Tried hylafax?****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus
> *Sent:* Tuesday, June 21, 2011 11:50 AM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] Fax API****
>
> ** **
>
> I've seen Faxing issues over and over in the channel, so since I don't need
> it so much I'd like to outsource it.****
>
> I've had a recommendation to use metrofax - they seem to have a nice
> website, great pricing, and an API that I may be able to get access to.***
> *
>
> Are there any other suggestions for a faxing both inbound and outbound that
> I can plug into my GUI for customers?****
>
> ** **
>
> Thanks!****
>
>
> ****
>
> -Avi Marcus****
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From bcxml at hotmail.com  Tue Jun 21 22:09:56 2011
From: bcxml at hotmail.com (Brian Campbell)
Date: Tue, 21 Jun 2011 14:09:56 -0400
Subject: [Freeswitch-users] Intermittent Audio Problem - Incomming Calls
In-Reply-To: <E811A8E4-8A8A-4EE3-BE8A-4A26B6E6C759@ipeva.fr>
References: <SNT135-w24A462340F95032961EE0FB7510@phx.gbl>, ,
	<EEBB17CE-15B0-4D72-A849-61505092D995@ipeva.fr>,
	<SNT135-w49311427DD49F93EF3280DB7510@phx.gbl>,
	<E811A8E4-8A8A-4EE3-BE8A-4A26B6E6C759@ipeva.fr>
Message-ID: <SNT135-w395A1BB72C59C5F214F609B7510@phx.gbl>



Thanks David
 
I have followed your instructions and I find that the audio is definately flowing between FreeSwitch and Speech Server on calls that seem to be broken. So that points me to a Speech Server issue
 
Thanks so much for the advice, you have been a big help
 
 
Brian Campbell
 
 



From: david.ponzone at ipeva.fr
Date: Tue, 21 Jun 2011 16:41:14 +0200
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Intermittent Audio Problem - Incomming Calls


Wireshark is your friend :)
Open your RTP flow, click on Telephony/RTP/Show All Streams
In the next window, select the stream you want (the one from FS to IVR) and click on Analyse.
And in the final window, click on Save Payload.
Then select the forward stream, and save as raw.


You should be able to import the raw file in any decent audio software (Audacity, etc...) as U-Law (I suppose you use PCMU), 8Khz.




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


Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr  -   www.ipeva-studio.com



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




Le 21/06/2011 ? 15:13, Brian Campbell a ?crit :


Thanks David

The Speech Server application reacts to speech from the caller validated by a grammar. There is a facility in Speech Server to log the calls, including the audio. On calls that work, I can hear the prompts from the application as well as my responses in the session audio. On calls that don't work, I hear the prompts from the application but none of my responses. After a small period of waiting the Speech Recognizer times out and tells me that it did not hear me say anything. I have tried increasing the timeout interval but the problem remains.
 
So I am not sure where the problem lies, is it something breaking down in Speech Server, or is the audio not making it from FreeSwitch to Speech Server. 
 
One thing for sure is that it doesnt stop working half way through the call. It either recognizes utterances from the caller or it doesn't, which would lead me to think that sometimes the call is not properly setup between Speech Server and FreeSwitch, but as I said I have run out of ideas. 
 
As I mentioned I do see RTP traffic, but I am afraid that I do not posses indepth knowledge about the protocol, so I am not sure how to properly analyse it.
 
Thanks again for your input
 
 
Brian Campbell
 



From: david.ponzone at ipeva.fr
Date: Tue, 21 Jun 2011 08:43:44 +0200
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Intermittent Audio Problem - Incomming Calls

Brian,


twice you say "the IVR doesn't seem to hear anything the caller is saying".
I suppose that means it is an assumption because it does not react to DTMFs ?
Do you have the possibility in this IVR to record the call ?
This way, you would be sure.


Also you say that Wireshark tells you the RTP traffic is fine.
Even when the call is not working ?
Do you notice anything different in this RTP ?
You may also take a RTP trace between FS and IVR of a non-working call, analyse the RTP flows and save the payload as raw to a file.
You should be able to listen to the audio with the right tool.




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


Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr  -   www.ipeva-studio.com



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




Le 21/06/2011 ? 04:29, Brian Campbell a ?crit :


I have run into a very strange issue with periodically losing Audio on Incomming calls
 
I am receiving incomming calls to FreeSwitch via a DID from an ITSP. Once FreeSwitch receives the call, I am adding a header to pass the UUID (which I use later to inject some custom values into CDRs) and then bridge to an IVR application running on Microsoft Speech Server 2007 which is on a separate box, using the following dial plan
 
<extension name="Inbound_FromSoTel">
  <condition field="destination_number" expression="^1(\d+)$">
    <action application="set" data="sip_h_X-FS_UUID=${uuid}" /> 
    <action application="bridge" data="sofia/external/$1 at 173.14.17.213:5060;transport=tcp" /> 
  </condition>
</extension>
 
Some calls are performing properly, but in some of the calls (about 30%) the IVR application doesn't seem to hear anything the caller is saying
 
I have eliminated any Firewalls for testing, so I go straight from the ITSP to FreeSwitch and then to Speech Server. Thee is no NAT in play.
 
There doesn't seem to be any rhyme or reason as to when a call is going to work, or when it isn't. I have also seen exactly the same behavior in a completely different environment with a different ITSP and a completely different IVR application.
 
So I figure that the problem is somewhere between FreeSwitch and Speech Server, but I don't see anything wrong. 
 
Using WireShark I can see the SIP signalling creating/tearing down the call and I see RTP traffic flowing both ways between FreeSwitch and Speech Server
 
The version of FreeSwitch that I am running is 'FreeSWITCH Version 1.0.head (git-38f06a3 2011-05-19 15-39-43 -0500)'
 
I have pasted 2 FreeSwitch traces into pastebin. The first one is of a working call, the second one is of a call where the IVR application doesn't seem to hear anything the caller is saying
 
Can anyone advise on what the issue might be. I am completely out of ideas.
 
Thanks
 
Brian Campbell

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

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


_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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


_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 		 	   		  
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From david.ponzone at ipeva.fr  Tue Jun 21 23:21:27 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 21 Jun 2011 21:21:27 +0200
Subject: [Freeswitch-users] Intermittent Audio Problem - Incomming Calls
In-Reply-To: <SNT135-w395A1BB72C59C5F214F609B7510@phx.gbl>
References: <SNT135-w24A462340F95032961EE0FB7510@phx.gbl>, ,
	<EEBB17CE-15B0-4D72-A849-61505092D995@ipeva.fr>,
	<SNT135-w49311427DD49F93EF3280DB7510@phx.gbl>,
	<E811A8E4-8A8A-4EE3-BE8A-4A26B6E6C759@ipeva.fr>
	<SNT135-w395A1BB72C59C5F214F609B7510@phx.gbl>
Message-ID: <54472BC3-EBC3-4BBF-8CFF-3C49F98C609C@ipeva.fr>

How surprising :)

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

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

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




Le 21/06/2011 ? 20:09, Brian Campbell a ?crit :

> 
> Thanks David
>  
> I have followed your instructions and I find that the audio is definately flowing between FreeSwitch and Speech Server on calls that seem to be broken. So that points me to a Speech Server issue
>  
> Thanks so much for the advice, you have been a big help
>  
>  
> Brian Campbell
>  
>  
> From: david.ponzone at ipeva.fr
> Date: Tue, 21 Jun 2011 16:41:14 +0200
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Intermittent Audio Problem - Incomming Calls
> 
> Wireshark is your friend :)
> Open your RTP flow, click on Telephony/RTP/Show All Streams
> In the next window, select the stream you want (the one from FS to IVR) and click on Analyse.
> And in the final window, click on Save Payload.
> Then select the forward stream, and save as raw.
> 
> You should be able to import the raw file in any decent audio software (Audacity, etc...) as U-Law (I suppose you use PCMU), 8Khz.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 21/06/2011 ? 15:13, Brian Campbell a ?crit :
> 
> Thanks David
> 
> The Speech Server application reacts to speech from the caller validated by a grammar. There is a facility in Speech Server to log the calls, including the audio. On calls that work, I can hear the prompts from the application as well as my responses in the session audio. On calls that don't work, I hear the prompts from the application but none of my responses. After a small period of waiting the Speech Recognizer times out and tells me that it did not hear me say anything. I have tried increasing the timeout interval but the problem remains.
>  
> So I am not sure where the problem lies, is it something breaking down in Speech Server, or is the audio not making it from FreeSwitch to Speech Server. 
>  
> One thing for sure is that it doesnt stop working half way through the call. It either recognizes utterances from the caller or it doesn't, which would lead me to think that sometimes the call is not properly setup between Speech Server and FreeSwitch, but as I said I have run out of ideas. 
>  
> As I mentioned I do see RTP traffic, but I am afraid that I do not posses indepth knowledge about the protocol, so I am not sure how to properly analyse it.
>  
> Thanks again for your input
>  
>  
> Brian Campbell
>  
> From: david.ponzone at ipeva.fr
> Date: Tue, 21 Jun 2011 08:43:44 +0200
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Intermittent Audio Problem - Incomming Calls
> 
> Brian,
> 
> twice you say "the IVR doesn't seem to hear anything the caller is saying".
> I suppose that means it is an assumption because it does not react to DTMFs ?
> Do you have the possibility in this IVR to record the call ?
> This way, you would be sure.
> 
> Also you say that Wireshark tells you the RTP traffic is fine.
> Even when the call is not working ?
> Do you notice anything different in this RTP ?
> You may also take a RTP trace between FS and IVR of a non-working call, analyse the RTP flows and save the payload as raw to a file.
> You should be able to listen to the audio with the right tool.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 21/06/2011 ? 04:29, Brian Campbell a ?crit :
> 
> I have run into a very strange issue with periodically losing Audio on Incomming calls
>  
> I am receiving incomming calls to FreeSwitch via a DID from an ITSP. Once FreeSwitch receives the call, I am adding a header to pass the UUID (which I use later to inject some custom values into CDRs) and then bridge to an IVR application running on Microsoft Speech Server 2007 which is on a separate box, using the following dial plan
>  
> <extension name="Inbound_FromSoTel">
>   <condition field="destination_number" expression="^1(\d+)$">
>     <action application="set" data="sip_h_X-FS_UUID=${uuid}" /> 
>     <action application="bridge" data="sofia/external/$1 at 173.14.17.213:5060;transport=tcp" /> 
>   </condition>
> </extension>
>  
> Some calls are performing properly, but in some of the calls (about 30%) the IVR application doesn't seem to hear anything the caller is saying
>  
> I have eliminated any Firewalls for testing, so I go straight from the ITSP to FreeSwitch and then to Speech Server. Thee is no NAT in play.
>  
> There doesn't seem to be any rhyme or reason as to when a call is going to work, or when it isn't. I have also seen exactly the same behavior in a completely different environment with a different ITSP and a completely different IVR application.
>  
> So I figure that the problem is somewhere between FreeSwitch and Speech Server, but I don't see anything wrong. 
>  
> Using WireShark I can see the SIP signalling creating/tearing down the call and I see RTP traffic flowing both ways between FreeSwitch and Speech Server
>  
> The version of FreeSwitch that I am running is 'FreeSWITCH Version 1.0.head (git-38f06a3 2011-05-19 15-39-43 -0500)'
>  
> I have pasted 2 FreeSwitch traces into pastebin. The first one is of a working call, the second one is of a call where the IVR application doesn't seem to hear anything the caller is saying
>  
> Can anyone advise on what the issue might be. I am completely out of ideas.
>  
> Thanks
>  
> Brian Campbell
> 
>  
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From mario_fs at mgtech.com  Wed Jun 22 03:28:50 2011
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 21 Jun 2011 16:28:50 -0700
Subject: [Freeswitch-users] FreeSWITCH Cookbook Recipe List
In-Reply-To: <BANLkTinuMMORNRSZPbF0MVVTUXugpMRCNA@mail.gmail.com>
References: <BANLkTima7xQPi2JYe908CfyZaYmOMsw=aQ@mail.gmail.com>
	<7EAFE942-322C-4EC0-941A-F1226B8DDAD1@mgtech.com>
	<BANLkTinuMMORNRSZPbF0MVVTUXugpMRCNA@mail.gmail.com>
Message-ID: <ABDCD6C9-316E-402E-9E80-FFCA8C53DC90@mgtech.com>

I get it, but beyond the install is how to control logging, setup emailing from FS, and a couple of other things that might be useful.

On Jun 20, 2011, at 8:26 AM, Michael Collins wrote:

> I believe we decided to forego the "how-to install fs" recipes since we have lots and lots of information on that subject on the wiki. We want to keep the recipes focused on interesting ways to use FreeSWITCH.
> 
> -MC
> 
> On Mon, Jun 20, 2011 at 7:55 AM, Mario G <mario_fs at mgtech.com> wrote:
> I see there is a need for install on OS X Server. I thought the OS X install I did for the wiki was quite comprehensive. What is missing? It should be the same for OS X server. Also, I will do a clean install on OS X 10.7 next month to update the wiki.
> 
> On Jun 17, 2011, at 8:14 PM, Michael Collins wrote:
> 
>> Hello all!
>> 
>> Some have inquired recently about what's needed to complete the FS cookbook. I've uploaded the most recent list of recipes that we have identified. Some are already written but many are yet to be done. Check out this page:
>> 
>> http://wiki.freeswitch.org/wiki/Cookbook#Information_for_prospective_authors
>> 
>> Contact me if you have any questions about helping with any of these recipes.
>> 
>> Thanks,
>> Michael
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jan.berger at video24.no  Wed Jun 22 03:36:00 2011
From: jan.berger at video24.no (Jan Berger)
Date: Wed, 22 Jun 2011 01:36:00 +0200
Subject: [Freeswitch-users] Microsoft Speech Server
Message-ID: <8C4E260F53CB47079D913881B45F6587@dell9400>

Hi,

 

Have anyone connected to Microsoft Speech Server from FreeSWITCH? 

 

I have briefly tested it on an Athlon II X2 215 2.7 Ghz. Windows 7 Home (not
with FS).

 

It works, but the CPU bounce quite high on a single channel - so I am
guessing 5 simultaneous voice streams before the CPU's are 100%. 

 

I am not a big fan of running speech on Windows, but I wanted to try it to
see if I can get it working.

 

Jan

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From eric at loopfx.com  Wed Jun 22 03:59:51 2011
From: eric at loopfx.com (Eric Beard)
Date: Tue, 21 Jun 2011 19:59:51 -0400
Subject: [Freeswitch-users] Microsoft Speech Server
In-Reply-To: <8C4E260F53CB47079D913881B45F6587@dell9400>
References: <8C4E260F53CB47079D913881B45F6587@dell9400>
Message-ID: <D4A666272A214C4DBBABFE569CA34CA401E025C00D@doctorbob.loop.local>

Yes, I am running it in production.

On a new Dell R610 (1U server) I can run 80 calls with CPU at maybe 5%, and I point several Speech Servers at a single Freeswitch box on a separate (Linux) server, also a 1U DELL.  That box can do more than 200 calls at around 10-20% CPU.

My calls are a mix of recorded messages and IVR.  Audio and reco quality are fine.

You could run Freeswitch on the same machine as Speech Server if you wanted.

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com<mailto:eric at loopfx.com>

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan Berger
Sent: Tuesday, June 21, 2011 7:36 PM
To: 'FreeSWITCH Users Help'
Subject: [Freeswitch-users] Microsoft Speech Server

Hi,

Have anyone connected to Microsoft Speech Server from FreeSWITCH?

I have briefly tested it on an Athlon II X2 215 2.7 Ghz. Windows 7 Home (not with FS).

It works, but the CPU bounce quite high on a single channel - so I am guessing 5 simultaneous voice streams before the CPU's are 100%.

I am not a big fan of running speech on Windows, but I wanted to try it to see if I can get it working.

Jan
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From jan.berger at video24.no  Wed Jun 22 04:20:05 2011
From: jan.berger at video24.no (Jan Berger)
Date: Wed, 22 Jun 2011 02:20:05 +0200
Subject: [Freeswitch-users] Microsoft Speech Server
In-Reply-To: <D4A666272A214C4DBBABFE569CA34CA401E025C00D@doctorbob.loop.local>
References: <8C4E260F53CB47079D913881B45F6587@dell9400>
	<D4A666272A214C4DBBABFE569CA34CA401E025C00D@doctorbob.loop.local>
Message-ID: <3851CB8FD5804BF9A30B87576460E4FE@dell9400>

How do you connect FreeSWITCH and SAPI?

 

I can't find anything on that topic either on FreeSWITCH or uniMRCP (I kind
of hoped uniMRCP would gateway into SAPI).

 

Jan

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Eric
Beard
Sent: 22. juni 2011 02:00
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Microsoft Speech Server

 

Yes, I am running it in production.

 

On a new Dell R610 (1U server) I can run 80 calls with CPU at maybe 5%, and
I point several Speech Servers at a single Freeswitch box on a separate
(Linux) server, also a 1U DELL.  That box can do more than 200 calls at
around 10-20% CPU.

 

My calls are a mix of recorded messages and IVR.  Audio and reco quality are
fine.

 

You could run Freeswitch on the same machine as Speech Server if you wanted.

 

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

Eric Z. Beard, CTO

Loop LLC

w (877) 850-2010 x9249

m (727) 776-2768

eric at loopfx.com

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan
Berger
Sent: Tuesday, June 21, 2011 7:36 PM
To: 'FreeSWITCH Users Help'
Subject: [Freeswitch-users] Microsoft Speech Server

 

Hi,

 

Have anyone connected to Microsoft Speech Server from FreeSWITCH? 

 

I have briefly tested it on an Athlon II X2 215 2.7 Ghz. Windows 7 Home (not
with FS).

 

It works, but the CPU bounce quite high on a single channel - so I am
guessing 5 simultaneous voice streams before the CPU's are 100%. 

 

I am not a big fan of running speech on Windows, but I wanted to try it to
see if I can get it working.

 

Jan

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From elijah at crankenstein.com  Wed Jun 22 04:44:48 2011
From: elijah at crankenstein.com (elijah)
Date: Tue, 21 Jun 2011 17:44:48 -0700
Subject: [Freeswitch-users] 'make current' fails to complete
Message-ID: <BANLkTikB19zJpVQSeXs8ouPN=zwdXy3zmQ@mail.gmail.com>

My attempts to 'make current' against the Git repository are failing today.
Can you recommend a method by which I can update my FreeSwitch installation?
Here's the the last bit of console logging:

...
make[2]: Leaving directory `/usr/src/freeswitch/libs/esl/ruby'
make -C java clean
make[2]: Entering directory `/usr/src/freeswitch/libs/esl/java'
rm -f *.o *.so *~ *.jar
make[2]: Leaving directory `/usr/src/freeswitch/libs/esl/java'
make -C managed clean
make[2]: Entering directory `/usr/src/freeswitch/libs/esl/managed'
rm -f *.o *.so *~
make[2]: Leaving directory `/usr/src/freeswitch/libs/esl/managed'
make[1]: Leaving directory `/usr/src/freeswitch/libs/esl'
make update
make[1]: Entering directory `/usr/src/freeswitch'
Pulling updates...
Updating 5923f71..4bb7683
src/mod/applications/mod_dptools/mod_dptools.c: needs update
fatal: Entry 'src/mod/applications/mod_dptools/mod_dptools.c' not uptodate.
Cannot merge.
make[1]: *** [update] Error 128
make[1]: Leaving directory `/usr/src/freeswitch'
make: *** [current] Error 2
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From curriegrad2004 at gmail.com  Wed Jun 22 05:11:58 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Tue, 21 Jun 2011 18:11:58 -0700
Subject: [Freeswitch-users] 'make current' fails to complete
In-Reply-To: <BANLkTikB19zJpVQSeXs8ouPN=zwdXy3zmQ@mail.gmail.com>
References: <BANLkTikB19zJpVQSeXs8ouPN=zwdXy3zmQ@mail.gmail.com>
Message-ID: <BANLkTin1FXE3KTtmUiYgQ-3qWiFOZkvQCg@mail.gmail.com>

did you make any changes to the code tree? If you did, you'd probably
want to perform a git stash to save all the changes you made then run
git stash apply to merge your changes after the pull.

Otherwise, it would be wise to run git reset --hard to 'fix' the
source tree if you are sure you made no changes to the code at all

On Tue, Jun 21, 2011 at 5:44 PM, elijah <elijah at crankenstein.com> wrote:
> My attempts to 'make current' against the Git repository are failing today.
> Can you?recommend?a method by which I can update my FreeSwitch installation?
> Here's the the last bit of console logging:
> ...
> make[2]: Leaving directory `/usr/src/freeswitch/libs/esl/ruby'
> make -C java clean
> make[2]: Entering directory `/usr/src/freeswitch/libs/esl/java'
> rm -f *.o *.so *~ *.jar
> make[2]: Leaving directory `/usr/src/freeswitch/libs/esl/java'
> make -C managed clean
> make[2]: Entering directory `/usr/src/freeswitch/libs/esl/managed'
> rm -f *.o *.so *~
> make[2]: Leaving directory `/usr/src/freeswitch/libs/esl/managed'
> make[1]: Leaving directory `/usr/src/freeswitch/libs/esl'
> make update
> make[1]: Entering directory `/usr/src/freeswitch'
> Pulling updates...
> Updating 5923f71..4bb7683
> src/mod/applications/mod_dptools/mod_dptools.c: needs update
> fatal: Entry 'src/mod/applications/mod_dptools/mod_dptools.c' not uptodate.
> Cannot merge.
> make[1]: *** [update] Error 128
> make[1]: Leaving directory `/usr/src/freeswitch'
> make: *** [current] Error 2
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From ankitwalia4u at gmail.com  Wed Jun 22 09:16:33 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Wed, 22 Jun 2011 10:46:33 +0530
Subject: [Freeswitch-users] Looking for Cost Effective IVR Solution for
	Concurrent Calls
Message-ID: <BANLkTi=TLxCRTxkJ-B5nO92YVcGOpBtp0A@mail.gmail.com>

Dear all,

I am working to develop an IVR solution for an NGO in Tamil Nadu, India
using Freeswitch open source telephony interface.

I am looking for any cost effective solution for hardware where concurrent
inbound/outbound calls can be made by the IVR system. Some research, I did
but I am skeptical, need some more guidance from experts. Number of
concurrent calls for initial stage inbound 5-10 but based on response from
the community, we will plan for scaling up. Outbound Scheduled calls are of
lesser importance right now.

I checked online that PRI line provides me 30 lines for voice which can be
used concurrently. But, the rent for the line is very high which cant be
feasible for the NGO.
Min PRI line cost is Rs 4.5 K per month.

Please suggest alternative solutions which I can use for the same.

1. What about VOIP  and using their local access number?
2. What about SIP based IP phone with extensions?
3. What about trunk lines, DID etc?
4. What about GSM Modems, GSMOpen?

I am very new telecom technology. Please help.

Thanks
Ankit

-- 
Life is like a rose its upto u feel it as its fragrance or thorns
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From sid.kshatriya at gmail.com  Wed Jun 22 09:59:47 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Wed, 22 Jun 2011 11:29:47 +0530
Subject: [Freeswitch-users] Looking for Cost Effective IVR Solution for
 Concurrent Calls
In-Reply-To: <BANLkTi=TLxCRTxkJ-B5nO92YVcGOpBtp0A@mail.gmail.com>
References: <BANLkTi=TLxCRTxkJ-B5nO92YVcGOpBtp0A@mail.gmail.com>
Message-ID: <BANLkTing4H1EJ+RawJ2d_REWmB_nUL8gFw@mail.gmail.com>

Any decent off the shelf desktop computer purchased today should be ok for
the kind of call volumes you're talking abnout.

You will need to buy a PRI data card which will cost Rs. 20-35 thousand
rupees. Look at Sangoma or Digium cards. I personally would recommend
Sangoma as seems to be better supported in freeswitch.

*Also look at kookoo.in -- you might be able to achieve what you want.*

I think the first thing you should do is understand the technology. After
you do that, you will be in a better position to ask questions and see
answers. Why do you look at reading a good book on freeswitch?

See more answers inline below....

On Wed, Jun 22, 2011 at 10:46 AM, ankIT WALiA <ankitwalia4u at gmail.com>wrote:

> Dear all,
>
> I am working to develop an IVR solution for an NGO in Tamil Nadu, India
> using Freeswitch open source telephony interface.
>
> I am looking for any cost effective solution for hardware where concurrent
> inbound/outbound calls can be made by the IVR system. Some research, I did
> but I am skeptical, need some more guidance from experts. Number of
> concurrent calls for initial stage inbound 5-10 but based on response from
> the community, we will plan for scaling up. Outbound Scheduled calls are of
> lesser importance right now.
>
> I checked online that PRI line provides me 30 lines for voice which can be
> used concurrently. But, the rent for the line is very high which cant be
> feasible for the NGO.
> Min PRI line cost is Rs 4.5 K per month.
>
> Please suggest alternative solutions which I can use for the same.
>
> 1. What about VOIP  and using their local access number?
>

VOIP is somewhat difficult to do in India. I don't know any services in
India that will give you a SIP connection.


> 2. What about SIP based IP phone with extensions?
>

You need to get a SIP connection or use a PRI.

>
> 3. What about trunk lines, DID etc?
>

You need to get a SIP connection or use a PRI


> 4. What about GSM Modems, GSMOpen?
>
>
May be an interesting option. But don't know much about it.


> I am very new telecom technology. Please help.
>
> Thanks
> Ankit
>
> --
> Life is like a rose its upto u feel it as its fragrance or thorns
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Wed Jun 22 12:40:42 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 22 Jun 2011 10:40:42 +0200
Subject: [Freeswitch-users] Minor issue with INVITE domain
Message-ID: <4BC0A7AA-CBC9-4A52-8C63-9C368F8D04EC@ipeva.fr>

Hello all,

I work with a provider which I need to send calls using their IP (I want to avoid any DNS issues), but I need the FQDN to be in the INVITE domain.
So basically, I have a gateway defined with
<param name="proxy" value="X.X.X.X"/>
but I need the INVITE to be:
sip:called at whatever.foo.com
and not:
sip:called at X.X.X.X

I played with everything I could think of:
outbound-proxy in gateway params (which seems to do nothing)

{sip_invite_domain=whatever.foo.com}bridge/gateway/gw/called
also does nothing.

I am missing something ?

Thanks

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

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

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




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From benkokakao at gmail.com  Wed Jun 22 12:42:36 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Wed, 22 Jun 2011 10:42:36 +0200
Subject: [Freeswitch-users] Modified Follow me
In-Reply-To: <4E00A61C.1060503@seletech.com>
References: <4E00A61C.1060503@seletech.com>
Message-ID: <BANLkTimSVMrTGJg72JJe5252z7J-RMXSLA@mail.gmail.com>

On 21 June 2011 16:09, Alessandro <a.luppi at seletech.com> wrote:
> Any suggestion?

While i can't give you a concrete solution, take a look at the files
that are created by fusionpbx, mainly
/usr/local/freeswitch/scripts/v_huntgroup_localhost_$EXTENSION.lua -
if you dig a bit you might be able to find and fix the problem
yourself(And eventually file a bugreport with fusionpbx).

hth
Christian


From sharad at coraltele.com  Wed Jun 22 13:17:53 2011
From: sharad at coraltele.com (sharad)
Date: Wed, 22 Jun 2011 14:47:53 +0530
Subject: [Freeswitch-users] Looking for Cost Effective IVR Solution
	forConcurrent Calls
References: <BANLkTi=TLxCRTxkJ-B5nO92YVcGOpBtp0A@mail.gmail.com>
Message-ID: <7091CBE8A30F42978D9F082C267C22DA@sharad>

Hi Ankit

I differ with you on the PRI pricing. Now a days, in north India, BSNL, Reliance, Airtel, etc, are providing PRI connectivity somewhere arround 10-12 K per month. More over this, they will permit you to make the free calls of the same value. Means if you opt for 10K / month plan, you will be able to make the out calls of wrth Rs 10K approx. Yes, if you want only inbound, you will have to pay at least 10K.  So you can check with the local service provider in Tamilnadu & nagotiate with the service provider for minimum rent. 

Regarding your rest queries, I am not sure what do you want to ask..plz explore your queries.

Regards
Sharad, Noida
9891499202


  ----- Original Message ----- 
  From: ankIT WALiA 
  To: FreeSWITCH Users Help ; The Linux-Delhi mailing list ; ilugc at ae.iitm.ac.in 
  Sent: Wednesday, June 22, 2011 10:46 AM
  Subject: [Freeswitch-users] Looking for Cost Effective IVR Solution forConcurrent Calls


  Dear all,


  I am working to develop an IVR solution for an NGO in Tamil Nadu, India using Freeswitch open source telephony interface. 


  I am looking for any cost effective solution for hardware where concurrent inbound/outbound calls can be made by the IVR system. Some research, I did but I am skeptical, need some more guidance from experts. Number of concurrent calls for initial stage inbound 5-10 but based on response from the community, we will plan for scaling up. Outbound Scheduled calls are of lesser importance right now.


  I checked online that PRI line provides me 30 lines for voice which can be used concurrently. But, the rent for the line is very high which cant be feasible for the NGO. 
  Min PRI line cost is Rs 4.5 K per month.


  Please suggest alternative solutions which I can use for the same. 


  1. What about VOIP  and using their local access number?
  2. What about SIP based IP phone with extensions?
  3. What about trunk lines, DID etc?
  4. What about GSM Modems, GSMOpen?


  I am very new telecom technology. Please help.


  Thanks
  Ankit

  -- 
  Life is like a rose its upto u feel it as its fragrance or thorns
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From ankitwalia4u at gmail.com  Wed Jun 22 13:48:20 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Wed, 22 Jun 2011 05:48:20 -0400
Subject: [Freeswitch-users] Looking for Cost Effective IVR Solution
 forConcurrent Calls
In-Reply-To: <7091CBE8A30F42978D9F082C267C22DA@sharad>
References: <BANLkTi=TLxCRTxkJ-B5nO92YVcGOpBtp0A@mail.gmail.com>
	<7091CBE8A30F42978D9F082C267C22DA@sharad>
Message-ID: <BANLkTikYip7UtFR1NBcooHGTGJwUHP1xDQ@mail.gmail.com>

I want to know If there is any other way to receive concurrent incomings
calls apart from using PRI.

I think Reliance is giving in 4.5K. I saw
herehttp://mobile.ebest.in/pri-tariff-plans/reliance-pri/3.html


On Wed, Jun 22, 2011 at 5:17 AM, sharad <sharad at coraltele.com> wrote:

> **
> Hi Ankit
>
> I differ with you on the PRI pricing. Now a days, in north India, BSNL,
> Reliance, Airtel, etc, are providing PRI connectivity somewhere arround
> 10-12 K per month. More over this, they will permit you to make the free
> calls of the same value. Means if you opt for 10K / month plan, you will be
> able to make the out calls of wrth Rs 10K approx. Yes, if you want only
> inbound, you will have to pay at least 10K.  So you can check with the local
> service provider in Tamilnadu & nagotiate with the service provider for
> minimum rent.
>
> Regarding your rest queries, I am not sure what do you want to ask..plz
> explore your queries.
>
> Regards
> Sharad, Noida
> 9891499202
>
>
>
> ----- Original Message -----
> *From:* ankIT WALiA <ankitwalia4u at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> ; The
> Linux-Delhi mailing list <ilugd at lists.linux-delhi.org> ;
> ilugc at ae.iitm.ac.in
> *Sent:* Wednesday, June 22, 2011 10:46 AM
> *Subject:* [Freeswitch-users] Looking for Cost Effective IVR Solution
> forConcurrent Calls
>
> Dear all,
>
> I am working to develop an IVR solution for an NGO in Tamil Nadu, India
> using Freeswitch open source telephony interface.
>
> I am looking for any cost effective solution for hardware where concurrent
> inbound/outbound calls can be made by the IVR system. Some research, I did
> but I am skeptical, need some more guidance from experts. Number of
> concurrent calls for initial stage inbound 5-10 but based on response from
> the community, we will plan for scaling up. Outbound Scheduled calls are of
> lesser importance right now.
>
> I checked online that PRI line provides me 30 lines for voice which can be
> used concurrently. But, the rent for the line is very high which cant be
> feasible for the NGO.
> Min PRI line cost is Rs 4.5 K per month.
>
> Please suggest alternative solutions which I can use for the same.
>
> 1. What about VOIP  and using their local access number?
> 2. What about SIP based IP phone with extensions?
> 3. What about trunk lines, DID etc?
> 4. What about GSM Modems, GSMOpen?
>
> I am very new telecom technology. Please help.
>
> Thanks
> Ankit
>
> --
> Life is like a rose its upto u feel it as its fragrance or thorns
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Life is like a rose its upto u feel it as its fragrance or thorns
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From david.ponzone at ipeva.fr  Wed Jun 22 14:45:58 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 22 Jun 2011 12:45:58 +0200
Subject: [Freeswitch-users] Looking for Cost Effective IVR Solution
	forConcurrent Calls
In-Reply-To: <BANLkTikYip7UtFR1NBcooHGTGJwUHP1xDQ@mail.gmail.com>
References: <BANLkTi=TLxCRTxkJ-B5nO92YVcGOpBtp0A@mail.gmail.com>
	<7091CBE8A30F42978D9F082C267C22DA@sharad>
	<BANLkTikYip7UtFR1NBcooHGTGJwUHP1xDQ@mail.gmail.com>
Message-ID: <C0329D33-AC4F-4281-838A-966C04C01250@ipeva.fr>

You could try to setup a SIP trunk with a foreign provider able to give you DIDs for India.
That's a long shot, and the quality could be at loss.

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

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

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




Le 22/06/2011 ? 11:48, ankIT WALiA a ?crit :

> I want to know If there is any other way to receive concurrent incomings calls apart from using PRI.
> 
> I think Reliance is giving in 4.5K. I saw here http://mobile.ebest.in/pri-tariff-plans/reliance-pri/3.html
> 
> 
> On Wed, Jun 22, 2011 at 5:17 AM, sharad <sharad at coraltele.com> wrote:
> Hi Ankit
>  
> I differ with you on the PRI pricing. Now a days, in north India, BSNL, Reliance, Airtel, etc, are providing PRI connectivity somewhere arround 10-12 K per month. More over this, they will permit you to make the free calls of the same value. Means if you opt for 10K / month plan, you will be able to make the out calls of wrth Rs 10K approx. Yes, if you want only inbound, you will have to pay at least 10K.  So you can check with the local service provider in Tamilnadu & nagotiate with the service provider for minimum rent.
>  
> Regarding your rest queries, I am not sure what do you want to ask..plz explore your queries.
>  
> Regards
> Sharad, Noida
> 9891499202
>  
>  
> ----- Original Message -----
> From: ankIT WALiA
> To: FreeSWITCH Users Help ; The Linux-Delhi mailing list ; ilugc at ae.iitm.ac.in
> Sent: Wednesday, June 22, 2011 10:46 AM
> Subject: [Freeswitch-users] Looking for Cost Effective IVR Solution forConcurrent Calls
> 
> Dear all,
> 
> I am working to develop an IVR solution for an NGO in Tamil Nadu, India using Freeswitch open source telephony interface. 
> 
> I am looking for any cost effective solution for hardware where concurrent inbound/outbound calls can be made by the IVR system. Some research, I did but I am skeptical, need some more guidance from experts. Number of concurrent calls for initial stage inbound 5-10 but based on response from the community, we will plan for scaling up. Outbound Scheduled calls are of lesser importance right now.
> 
> I checked online that PRI line provides me 30 lines for voice which can be used concurrently. But, the rent for the line is very high which cant be feasible for the NGO. 
> Min PRI line cost is Rs 4.5 K per month.
> 
> Please suggest alternative solutions which I can use for the same. 
> 
> 1. What about VOIP  and using their local access number?
> 2. What about SIP based IP phone with extensions?
> 3. What about trunk lines, DID etc?
> 4. What about GSM Modems, GSMOpen?
> 
> I am very new telecom technology. Please help.
> 
> Thanks
> Ankit
> 
> -- 
> Life is like a rose its upto u feel it as its fragrance or thorns
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Life is like a rose its upto u feel it as its fragrance or thorns
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From sos at sokhapkin.dyndns.org  Wed Jun 22 15:51:25 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 22 Jun 2011 07:51:25 -0400
Subject: [Freeswitch-users] Minor issue with INVITE domain
In-Reply-To: <4BC0A7AA-CBC9-4A52-8C63-9C368F8D04EC@ipeva.fr>
References: <4BC0A7AA-CBC9-4A52-8C63-9C368F8D04EC@ipeva.fr>
Message-ID: <201106220751.25728.sos@sokhapkin.dyndns.org>

Put IP address of whatever.foo.com to /etc/hosts and use domain name. you will 
not get any DNS issues.

On Wednesday 22 June 2011, David Ponzone wrote:
> Hello all,
> 
> I work with a provider which I need to send calls using their IP (I want to
> avoid any DNS issues), but I need the FQDN to be in the INVITE domain. So
> basically, I have a gateway defined with
> <param name="proxy" value="X.X.X.X"/>
> but I need the INVITE to be:
> sip:called at whatever.foo.com
> and not:
> sip:called at X.X.X.X
> 
> I played with everything I could think of:
> outbound-proxy in gateway params (which seems to do nothing)
> 
> {sip_invite_domain=whatever.foo.com}bridge/gateway/gw/called
> also does nothing.
> 
> I am missing something ?
> 
> Thanks
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
> s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de
> ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.



From sharad at coraltele.com  Wed Jun 22 16:01:14 2011
From: sharad at coraltele.com (sharad)
Date: Wed, 22 Jun 2011 17:31:14 +0530
Subject: [Freeswitch-users] Looking for Cost Effective IVR
	SolutionforConcurrent Calls
References: <BANLkTi=TLxCRTxkJ-B5nO92YVcGOpBtp0A@mail.gmail.com><7091CBE8A30F42978D9F082C267C22DA@sharad>
	<BANLkTikYip7UtFR1NBcooHGTGJwUHP1xDQ@mail.gmail.com>
Message-ID: <35A852F7A200457DA761EBAEDC836F23@sharad>

Yes, there are following ways -

1.    Analog trunk line (FXO) - taken from BSNL. Connect those lines to FXO GW / PBX & route the calls to freeswitch. If you are expecting your incoming calls within India, this may be cheaper.

2.    PRI - You already know.

3.    Through VoIP - If you are expecting international calls, this medium may be cheaper. But you need to discuss this with your VoIP provider in advance that you r expecting incoming calls also.

Regards
Sharad


  ----- Original Message ----- 
  From: ankIT WALiA 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, June 22, 2011 3:18 PM
  Subject: Re: [Freeswitch-users] Looking for Cost Effective IVR SolutionforConcurrent Calls


  I want to know If there is any other way to receive concurrent incomings calls apart from using PRI.

  I think Reliance is giving in 4.5K. I saw here http://mobile.ebest.in/pri-tariff-plans/reliance-pri/3.html



  On Wed, Jun 22, 2011 at 5:17 AM, sharad <sharad at coraltele.com> wrote:

    Hi Ankit

    I differ with you on the PRI pricing. Now a days, in north India, BSNL, Reliance, Airtel, etc, are providing PRI connectivity somewhere arround 10-12 K per month. More over this, they will permit you to make the free calls of the same value. Means if you opt for 10K / month plan, you will be able to make the out calls of wrth Rs 10K approx. Yes, if you want only inbound, you will have to pay at least 10K.  So you can check with the local service provider in Tamilnadu & nagotiate with the service provider for minimum rent. 

    Regarding your rest queries, I am not sure what do you want to ask..plz explore your queries.

    Regards
    Sharad, Noida
    9891499202


      ----- Original Message ----- 
      From: ankIT WALiA 
      To: FreeSWITCH Users Help ; The Linux-Delhi mailing list ; ilugc at ae.iitm.ac.in 
      Sent: Wednesday, June 22, 2011 10:46 AM
      Subject: [Freeswitch-users] Looking for Cost Effective IVR Solution forConcurrent Calls


      Dear all, 


      I am working to develop an IVR solution for an NGO in Tamil Nadu, India using Freeswitch open source telephony interface. 


      I am looking for any cost effective solution for hardware where concurrent inbound/outbound calls can be made by the IVR system. Some research, I did but I am skeptical, need some more guidance from experts. Number of concurrent calls for initial stage inbound 5-10 but based on response from the community, we will plan for scaling up. Outbound Scheduled calls are of lesser importance right now.


      I checked online that PRI line provides me 30 lines for voice which can be used concurrently. But, the rent for the line is very high which cant be feasible for the NGO. 
      Min PRI line cost is Rs 4.5 K per month.


      Please suggest alternative solutions which I can use for the same. 


      1. What about VOIP  and using their local access number?
      2. What about SIP based IP phone with extensions?
      3. What about trunk lines, DID etc?
      4. What about GSM Modems, GSMOpen?


      I am very new telecom technology. Please help.


      Thanks
      Ankit

      -- 
      Life is like a rose its upto u feel it as its fragrance or thorns


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

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





  -- 
  Life is like a rose its upto u feel it as its fragrance or thorns
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From freeswitch at mralston.com  Wed Jun 22 16:38:32 2011
From: freeswitch at mralston.com (Matthew Ralston)
Date: Wed, 22 Jun 2011 13:38:32 +0100
Subject: [Freeswitch-users] Call drops after 30 seconds
Message-ID: <71F1700A-7641-43E5-A7C6-F25A0DC2EBC5@mralston.com>

Hi,

I'm having a problem at the moment with calls being successfully set up, with two-way audio, being terminated by FreeSWITCH after 30 seconds.

Internal calls (i.e. between SIP phones on the same LAN segment as the FreeSWITCH box) work flawlessly.

The problem arises when at least one of the handsets is located elsewhere on the Internet. This behaviour is exhibited under the following circumstances:

- A-leg only call, e.g. to voicemail when the handset is at another location on the Internet
- A-leg-B-leg call if one or both of the handsets are at another location on the Internet
- Inbound calls from our external SIP provider
- Outbound calls to our external SIP provider

So it is obvious that the problem is related to the SIP going via the Internet, but I'm having trouble understanding why.

Whilst debugging this problem I have placed the FreeSWITCH box is in the DMZ on our router, so there should not be any ports blocked. The FreeSWITCH box itself is not running a software firewall.

The calls themselves are absolutely fine for the first 30 seconds - each party can hear the other talking fine.

The fact that the call is consistently dropped after 30 seconds (give or take a second or two for PDD) suggests that some timeout is being triggered.

When FreeSWITCH terminates the call, the following is logged to the console:

2011-06-22 13:33:50.514941 [DEBUG] sofia.c:4787 Channel sofia/internal/1006 at public.ip.removed entering state [terminating][0]
2011-06-22 13:33:50.514941 [DEBUG] switch_channel.c:2641 (sofia/internal/1006 at public.ip.removed) Callstate Change ACTIVE -> HANGUP
2011-06-22 13:33:50.514941 [NOTICE] sofia.c:5508 Hangup sofia/internal/1006 at public.ip.removed [CS_EXECUTE] [NORMAL_UNSPECIFIED]
2011-06-22 13:33:50.514941 [DEBUG] switch_channel.c:2657 Send signal sofia/internal/1006 at public.ip.removed [KILL]
2011-06-22 13:33:50.514941 [DEBUG] switch_core_session.c:1118 Send signal sofia/internal/1006 at public.ip.removed [BREAK]
2011-06-22 13:33:50.534966 [DEBUG] switch_ivr_play_say.c:1649 done playing file
2011-06-22 13:33:50.625988 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-press.wav] (en:en)
2011-06-22 13:33:50.727027 [DEBUG] switch_core_session.c:2063 sofia/internal/1006 at public.ip.removed skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:371 (sofia/internal/1006 at public.ip.removed) State EXECUTE going to sleep
2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1006 at public.ip.removed) Running State Change CS_HANGUP
2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/1006 at public.ip.removed) State HANGUP
2011-06-22 13:33:50.727027 [DEBUG] mod_sofia.c:458 Channel sofia/internal/1006 at public.ip.removed hanging up, cause: NORMAL_UNSPECIFIED
2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1006 at public.ip.removed Standard HANGUP, cause: NORMAL_UNSPECIFIED
2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/1006 at public.ip.removed) State HANGUP going to sleep
2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/1006 at public.ip.removed) State Change CS_HANGUP -> CS_REPORTING
2011-06-22 13:33:50.727027 [DEBUG] switch_core_session.c:1118 Send signal sofia/internal/1006 at public.ip.removed [BREAK]
2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1006 at public.ip.removed) Running State Change CS_REPORTING
2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/1006 at public.ip.removed) State REPORTING
2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1006 at public.ip.removed Standard REPORTING, cause: NORMAL_UNSPECIFIED
2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/1006 at public.ip.removed) State REPORTING going to sleep
2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:350 (sofia/internal/1006 at public.ip.removed) State Change CS_REPORTING -> CS_DESTROY
2011-06-22 13:33:50.740064 [DEBUG] switch_core_session.c:1118 Send signal sofia/internal/1006 at public.ip.removed [BREAK]
2011-06-22 13:33:50.740064 [DEBUG] switch_core_session.c:1290 Session 5 (sofia/internal/1006 at public.ip.removed) Locked, Waiting on external entities
2011-06-22 13:33:50.740064 [NOTICE] switch_core_session.c:1308 Session 5 (sofia/internal/1006 at public.ip.removed) Ended
2011-06-22 13:33:50.740064 [NOTICE] switch_core_session.c:1310 Close Channel sofia/internal/1006 at public.ip.removed [CS_DESTROY]
2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/1006 at public.ip.removed) Callstate Change HANGUP -> DOWN
2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:457 (sofia/internal/1006 at public.ip.removed) Running State Change CS_DESTROY
2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1006 at public.ip.removed) State DESTROY
2011-06-22 13:33:50.740064 [DEBUG] mod_sofia.c:363 sofia/internal/1006 at public.ip.removed SOFIA DESTROY
2011-06-22 13:33:50.780056 [DEBUG] switch_nat.c:570 unmapped public port 31484 protocol UDP to localport 31484
2011-06-22 13:33:50.840070 [DEBUG] switch_nat.c:570 unmapped public port 31485 protocol UDP to localport 31485
2011-06-22 13:33:50.840070 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1006 at public.ip.removed Standard DESTROY
2011-06-22 13:33:50.840070 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1006 at public.ip.removed) State DESTROY going to sleep

The above example was from an externally situated SIP phone ringing voicemail (4000) on FreeSWITCH.

I have experimented changing various timers and timeouts in the config of FreeSWITCH (one at a time, being careful to put them back afterwards!) but been unable to resolve the issue.

Incidentally, we have no long term intention of running off-site SIP phones with the PBX and I'm hoping not to have to leave it in the DMZ either, it's just like that for debugging. What is a real issue is the calls to our external SIP provider (i.e. outbound calls) being dropped.

Any suggestions would be greatly appreciated.

Thanks,

Matthew Ralston
Web Developer & IT Consultant

matt at mralston.co.uk
www.mralston.com



From mattzerah+freeswitch at gmail.com  Wed Jun 22 16:44:15 2011
From: mattzerah+freeswitch at gmail.com (Matt Paine)
Date: Wed, 22 Jun 2011 22:44:15 +1000
Subject: [Freeswitch-users] Call drops after 30 seconds
In-Reply-To: <71F1700A-7641-43E5-A7C6-F25A0DC2EBC5@mralston.com>
References: <71F1700A-7641-43E5-A7C6-F25A0DC2EBC5@mralston.com>
Message-ID: <BANLkTi=_xqE4WiGn1CqoofbS-9_txzqGkw@mail.gmail.com>

My two cents worth...

I have fixed this in the past by setting up a STUND server to help with the
SIP data. I had the same problem as you, 30 seconds on the dot the call
would drop. After pointing my sip devices to a stun server everything is
solved. (for me anyway :)

HTH

Matt


On 22 June 2011 22:38, Matthew Ralston <freeswitch at mralston.com> wrote:

> Hi,
>
> I'm having a problem at the moment with calls being successfully set up,
> with two-way audio, being terminated by FreeSWITCH after 30 seconds.
>
> Internal calls (i.e. between SIP phones on the same LAN segment as the
> FreeSWITCH box) work flawlessly.
>
> The problem arises when at least one of the handsets is located elsewhere
> on the Internet. This behaviour is exhibited under the following
> circumstances:
>
> - A-leg only call, e.g. to voicemail when the handset is at another
> location on the Internet
> - A-leg-B-leg call if one or both of the handsets are at another location
> on the Internet
> - Inbound calls from our external SIP provider
> - Outbound calls to our external SIP provider
>
> So it is obvious that the problem is related to the SIP going via the
> Internet, but I'm having trouble understanding why.
>
> Whilst debugging this problem I have placed the FreeSWITCH box is in the
> DMZ on our router, so there should not be any ports blocked. The FreeSWITCH
> box itself is not running a software firewall.
>
> The calls themselves are absolutely fine for the first 30 seconds - each
> party can hear the other talking fine.
>
> The fact that the call is consistently dropped after 30 seconds (give or
> take a second or two for PDD) suggests that some timeout is being triggered.
>
> When FreeSWITCH terminates the call, the following is logged to the
> console:
>
> 2011-06-22 13:33:50.514941 [DEBUG] sofia.c:4787 Channel
> sofia/internal/1006 at public.ip.removed entering state [terminating][0]
> 2011-06-22 13:33:50.514941 [DEBUG] switch_channel.c:2641
> (sofia/internal/1006 at public.ip.removed) Callstate Change ACTIVE -> HANGUP
> 2011-06-22 13:33:50.514941 [NOTICE] sofia.c:5508 Hangup
> sofia/internal/1006 at public.ip.removed [CS_EXECUTE] [NORMAL_UNSPECIFIED]
> 2011-06-22 13:33:50.514941 [DEBUG] switch_channel.c:2657 Send signal
> sofia/internal/1006 at public.ip.removed [KILL]
> 2011-06-22 13:33:50.514941 [DEBUG] switch_core_session.c:1118 Send signal
> sofia/internal/1006 at public.ip.removed [BREAK]
> 2011-06-22 13:33:50.534966 [DEBUG] switch_ivr_play_say.c:1649 done playing
> file
> 2011-06-22 13:33:50.625988 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-press.wav] (en:en)
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_session.c:2063
> sofia/internal/1006 at public.ip.removed skip receive message
> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:371
> (sofia/internal/1006 at public.ip.removed) State EXECUTE going to sleep
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:325
> (sofia/internal/1006 at public.ip.removed) Running State Change CS_HANGUP
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:565
> (sofia/internal/1006 at public.ip.removed) State HANGUP
> 2011-06-22 13:33:50.727027 [DEBUG] mod_sofia.c:458 Channel
> sofia/internal/1006 at public.ip.removed hanging up, cause:
> NORMAL_UNSPECIFIED
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/1006 at public.ip.removed Standard HANGUP, cause:
> NORMAL_UNSPECIFIED
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:565
> (sofia/internal/1006 at public.ip.removed) State HANGUP going to sleep
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:356
> (sofia/internal/1006 at public.ip.removed) State Change CS_HANGUP ->
> CS_REPORTING
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_session.c:1118 Send signal
> sofia/internal/1006 at public.ip.removed [BREAK]
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:325
> (sofia/internal/1006 at public.ip.removed) Running State Change CS_REPORTING
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:625
> (sofia/internal/1006 at public.ip.removed) State REPORTING
> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/1006 at public.ip.removed Standard REPORTING, cause:
> NORMAL_UNSPECIFIED
> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:625
> (sofia/internal/1006 at public.ip.removed) State REPORTING going to sleep
> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:350
> (sofia/internal/1006 at public.ip.removed) State Change CS_REPORTING ->
> CS_DESTROY
> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_session.c:1118 Send signal
> sofia/internal/1006 at public.ip.removed [BREAK]
> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_session.c:1290 Session 5
> (sofia/internal/1006 at public.ip.removed) Locked, Waiting on external
> entities
> 2011-06-22 13:33:50.740064 [NOTICE] switch_core_session.c:1308 Session 5
> (sofia/internal/1006 at public.ip.removed) Ended
> 2011-06-22 13:33:50.740064 [NOTICE] switch_core_session.c:1310 Close
> Channel sofia/internal/1006 at public.ip.removed [CS_DESTROY]
> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:454
> (sofia/internal/1006 at public.ip.removed) Callstate Change HANGUP -> DOWN
> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:457
> (sofia/internal/1006 at public.ip.removed) Running State Change CS_DESTROY
> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:467
> (sofia/internal/1006 at public.ip.removed) State DESTROY
> 2011-06-22 13:33:50.740064 [DEBUG] mod_sofia.c:363
> sofia/internal/1006 at public.ip.removed SOFIA DESTROY
> 2011-06-22 13:33:50.780056 [DEBUG] switch_nat.c:570 unmapped public port
> 31484 protocol UDP to localport 31484
> 2011-06-22 13:33:50.840070 [DEBUG] switch_nat.c:570 unmapped public port
> 31485 protocol UDP to localport 31485
> 2011-06-22 13:33:50.840070 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/1006 at public.ip.removed Standard DESTROY
> 2011-06-22 13:33:50.840070 [DEBUG] switch_core_state_machine.c:467
> (sofia/internal/1006 at public.ip.removed) State DESTROY going to sleep
>
> The above example was from an externally situated SIP phone ringing
> voicemail (4000) on FreeSWITCH.
>
> I have experimented changing various timers and timeouts in the config of
> FreeSWITCH (one at a time, being careful to put them back afterwards!) but
> been unable to resolve the issue.
>
> Incidentally, we have no long term intention of running off-site SIP phones
> with the PBX and I'm hoping not to have to leave it in the DMZ either, it's
> just like that for debugging. What is a real issue is the calls to our
> external SIP provider (i.e. outbound calls) being dropped.
>
> Any suggestions would be greatly appreciated.
>
> Thanks,
>
> Matthew Ralston
> Web Developer & IT Consultant
>
> matt at mralston.co.uk
> www.mralston.com
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From hemant.taan at gmail.com  Wed Jun 22 16:54:39 2011
From: hemant.taan at gmail.com (hemant kumar)
Date: Wed, 22 Jun 2011 18:24:39 +0530
Subject: [Freeswitch-users] Looking for Cost Effective IVR Solution for
 Concurrent Calls
Message-ID: <BANLkTikS6ayTx+B0i8WNGB0U6KFhKSEDdw@mail.gmail.com>

You should be able to get pri with min rent of 1750rs from BSNL as i
had got one installed myself, after the whatever is the usage.

On Wednesday, June 22, 2011, sharad <sharad at coraltele.com> wrote:
>
>
>
>
>
>
>
> Yes, there are following?ways
> -
>
> 1.??? Analog trunk line
> (FXO) - taken from BSNL. Connect those lines to FXO GW / PBX & route the
> calls to freeswitch. If you are expecting your incoming calls within India, this
> may be cheaper.
>
> 2.??? PRI - You already
> know.
>
> 3.??? Through VoIP - If
> you are expecting international calls, this medium may be cheaper. But you need
> to discuss this with your VoIP provider in advance that you r expecting incoming
> calls also.
>
> Regards
> Sharad
>
>
>
>   ----- Original Message -----
>   From:
>   ankIT
>   WALiA?<ankitwalia4u at gmail.com>
>   To: FreeSWITCH Users Help?<freeswitch-users at lists.freeswitch.org>
>
>   Sent: Wednesday, June 22, 2011 3:18
>   PM
>   Subject: Re: [Freeswitch-users] Looking
>   for Cost Effective IVR SolutionforConcurrent Calls
>
> I want to know If there is any other way to receive concurrent
>   incomings calls apart from using PRI.
>
> I think Reliance is giving in
>   4.5K. I saw here
>   http://mobile.ebest.in/pri-tariff-plans/reliance-pri/3.html
>
>
>   On Wed, Jun 22, 2011 at 5:17 AM, sharad <sharad at coraltele.com>
>   wrote:
>
>
>     Hi Ankit
>
>     I differ with you on the PRI pricing.
>     Now a days, in north India, BSNL, Reliance,?Airtel, etc, are providing
>     PRI connectivity somewhere arround 10-12 K per month. More over this, they
>     will permit you to make the free calls of the same value. Means if you opt
>     for 10K / month plan, you will be able to make the out calls of wrth Rs 10K
>     approx. Yes, if you want only inbound, you will have to pay at least
>     10K.? So you can check with the local service provider in Tamilnadu
>     & nagotiate with the service provider for minimum rent.
>
>     Regarding your?rest queries, I
>     am not sure what do you want to ask..plz explore your queries.
>
>     Regards
>     Sharad, Noida
>     9891499202
>
>
>
>
>
>
>       ----- Original Message -----
>       From: ankIT WALiA
>       To: FreeSWITCH Users Help ; The Linux-Delhi
>       mailing list ; ilugc at ae.iitm.ac.in
>
>       Sent: Wednesday, June 22, 2011 10:46
>       AM
>       Subject: [Freeswitch-users] Looking
>       for Cost Effective IVR Solution forConcurrent Calls
>
> Dear all,
>
>
>       I am working to develop an IVR solution for an NGO in Tamil Nadu,
>       India using Freeswitch open source telephony interface.
>
>
>       I am looking for any cost effective solution for hardware where
>       concurrent inbound/outbound calls can be made by the IVR system.?Some
>       research, I did but I am skeptical, need some more guidance from experts.
>       Number of concurrent calls for initial stage inbound 5-10 but based on
>       response from the community, we will plan for scaling up. Outbound
>       Scheduled calls are of lesser?importance right now.
>
>
>       I checked online that PRI line provides me 30 lines for voice which
>       can be used concurrently. But, the rent for the line is very high which
>       cant be feasible for the NGO.
>       Min PRI line cost is Rs 4.5 K per month.
>
>
>       Please suggest alternative solutions which I can use for the
>       same.
>
>
>       1. What about VOIP ?and using their local access number?
>       2. What about SIP based IP phone with extensions?
>       3. What about trunk lines, DID etc?
>       4. What about GSM Modems, GSMOpen?
>
>
>       I am very new telecom technology. Please help.
>
>
>       Thanks
>       Ankit
>
> --
> Life is
>


From eric at loopfx.com  Wed Jun 22 17:00:29 2011
From: eric at loopfx.com (Eric Beard)
Date: Wed, 22 Jun 2011 09:00:29 -0400
Subject: [Freeswitch-users] Microsoft Speech Server
In-Reply-To: <3851CB8FD5804BF9A30B87576460E4FE@dell9400>
References: <8C4E260F53CB47079D913881B45F6587@dell9400>
	<D4A666272A214C4DBBABFE569CA34CA401E025C00D@doctorbob.loop.local>
	<3851CB8FD5804BF9A30B87576460E4FE@dell9400>
Message-ID: <D4A666272A214C4DBBABFE569CA34CA401E025C044@doctorbob.loop.local>

Not sure about SAPI.  I just set the trusted sip peer to point to Freeswitch.

There's a forum here that might help:

http://gotspeech.net/forums/thread/10657.aspx

and a series of blog posts here:

http://gotspeech.net/blogs/verbalinput/archive/2010/11/12/speech-server-2007-marries-freeswitch-part-6-upgrading-to-1-0-6.aspx



-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com<mailto:eric at loopfx.com>

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan Berger
Sent: Tuesday, June 21, 2011 8:20 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] Microsoft Speech Server

How do you connect FreeSWITCH and SAPI?

I can't find anything on that topic either on FreeSWITCH or uniMRCP (I kind of hoped uniMRCP would gateway into SAPI).

Jan
________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Eric Beard
Sent: 22. juni 2011 02:00
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Microsoft Speech Server

Yes, I am running it in production.

On a new Dell R610 (1U server) I can run 80 calls with CPU at maybe 5%, and I point several Speech Servers at a single Freeswitch box on a separate (Linux) server, also a 1U DELL.  That box can do more than 200 calls at around 10-20% CPU.

My calls are a mix of recorded messages and IVR.  Audio and reco quality are fine.

You could run Freeswitch on the same machine as Speech Server if you wanted.

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com<mailto:eric at loopfx.com>

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan Berger
Sent: Tuesday, June 21, 2011 7:36 PM
To: 'FreeSWITCH Users Help'
Subject: [Freeswitch-users] Microsoft Speech Server

Hi,

Have anyone connected to Microsoft Speech Server from FreeSWITCH?

I have briefly tested it on an Athlon II X2 215 2.7 Ghz. Windows 7 Home (not with FS).

It works, but the CPU bounce quite high on a single channel - so I am guessing 5 simultaneous voice streams before the CPU's are 100%.

I am not a big fan of running speech on Windows, but I wanted to try it to see if I can get it working.

Jan
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From chris.chen2004 at gmail.com  Wed Jun 22 17:21:37 2011
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Wed, 22 Jun 2011 09:21:37 -0400
Subject: [Freeswitch-users] Call drops after 30 seconds
In-Reply-To: <71F1700A-7641-43E5-A7C6-F25A0DC2EBC5@mralston.com>
References: <71F1700A-7641-43E5-A7C6-F25A0DC2EBC5@mralston.com>
Message-ID: <BANLkTikoP+hvtbzDn6fm_O6MmSN--x7TyA@mail.gmail.com>

Hi Matthew, this is typical behavior for the setup of SIP behind NAT.

    1) Please provide the exact setup of remote SIP phones, what's the
router model, does it have SIP ALG enabled, what kind  of SIP phones
    2)


On Wed, Jun 22, 2011 at 8:38 AM, Matthew Ralston <freeswitch at mralston.com>wrote:

> Hi,
>
> I'm having a problem at the moment with calls being successfully set up,
> with two-way audio, being terminated by FreeSWITCH after 30 seconds.
>
> Internal calls (i.e. between SIP phones on the same LAN segment as the
> FreeSWITCH box) work flawlessly.
>
> The problem arises when at least one of the handsets is located elsewhere
> on the Internet. This behaviour is exhibited under the following
> circumstances:
>
> - A-leg only call, e.g. to voicemail when the handset is at another
> location on the Internet
> - A-leg-B-leg call if one or both of the handsets are at another location
> on the Internet
> - Inbound calls from our external SIP provider
> - Outbound calls to our external SIP provider
>
> So it is obvious that the problem is related to the SIP going via the
> Internet, but I'm having trouble understanding why.
>
> Whilst debugging this problem I have placed the FreeSWITCH box is in the
> DMZ on our router, so there should not be any ports blocked. The FreeSWITCH
> box itself is not running a software firewall.
>
> The calls themselves are absolutely fine for the first 30 seconds - each
> party can hear the other talking fine.
>
> The fact that the call is consistently dropped after 30 seconds (give or
> take a second or two for PDD) suggests that some timeout is being triggered.
>
> When FreeSWITCH terminates the call, the following is logged to the
> console:
>
> 2011-06-22 13:33:50.514941 [DEBUG] sofia.c:4787 Channel
> sofia/internal/1006 at public.ip.removed entering state [terminating][0]
> 2011-06-22 13:33:50.514941 [DEBUG] switch_channel.c:2641
> (sofia/internal/1006 at public.ip.removed) Callstate Change ACTIVE -> HANGUP
> 2011-06-22 13:33:50.514941 [NOTICE] sofia.c:5508 Hangup
> sofia/internal/1006 at public.ip.removed [CS_EXECUTE] [NORMAL_UNSPECIFIED]
> 2011-06-22 13:33:50.514941 [DEBUG] switch_channel.c:2657 Send signal
> sofia/internal/1006 at public.ip.removed [KILL]
> 2011-06-22 13:33:50.514941 [DEBUG] switch_core_session.c:1118 Send signal
> sofia/internal/1006 at public.ip.removed [BREAK]
> 2011-06-22 13:33:50.534966 [DEBUG] switch_ivr_play_say.c:1649 done playing
> file
> 2011-06-22 13:33:50.625988 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-press.wav] (en:en)
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_session.c:2063
> sofia/internal/1006 at public.ip.removed skip receive message
> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:371
> (sofia/internal/1006 at public.ip.removed) State EXECUTE going to sleep
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:325
> (sofia/internal/1006 at public.ip.removed) Running State Change CS_HANGUP
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:565
> (sofia/internal/1006 at public.ip.removed) State HANGUP
> 2011-06-22 13:33:50.727027 [DEBUG] mod_sofia.c:458 Channel
> sofia/internal/1006 at public.ip.removed hanging up, cause:
> NORMAL_UNSPECIFIED
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/1006 at public.ip.removed Standard HANGUP, cause:
> NORMAL_UNSPECIFIED
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:565
> (sofia/internal/1006 at public.ip.removed) State HANGUP going to sleep
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:356
> (sofia/internal/1006 at public.ip.removed) State Change CS_HANGUP ->
> CS_REPORTING
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_session.c:1118 Send signal
> sofia/internal/1006 at public.ip.removed [BREAK]
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:325
> (sofia/internal/1006 at public.ip.removed) Running State Change CS_REPORTING
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:625
> (sofia/internal/1006 at public.ip.removed) State REPORTING
> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/1006 at public.ip.removed Standard REPORTING, cause:
> NORMAL_UNSPECIFIED
> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:625
> (sofia/internal/1006 at public.ip.removed) State REPORTING going to sleep
> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:350
> (sofia/internal/1006 at public.ip.removed) State Change CS_REPORTING ->
> CS_DESTROY
> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_session.c:1118 Send signal
> sofia/internal/1006 at public.ip.removed [BREAK]
> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_session.c:1290 Session 5
> (sofia/internal/1006 at public.ip.removed) Locked, Waiting on external
> entities
> 2011-06-22 13:33:50.740064 [NOTICE] switch_core_session.c:1308 Session 5
> (sofia/internal/1006 at public.ip.removed) Ended
> 2011-06-22 13:33:50.740064 [NOTICE] switch_core_session.c:1310 Close
> Channel sofia/internal/1006 at public.ip.removed [CS_DESTROY]
> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:454
> (sofia/internal/1006 at public.ip.removed) Callstate Change HANGUP -> DOWN
> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:457
> (sofia/internal/1006 at public.ip.removed) Running State Change CS_DESTROY
> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:467
> (sofia/internal/1006 at public.ip.removed) State DESTROY
> 2011-06-22 13:33:50.740064 [DEBUG] mod_sofia.c:363
> sofia/internal/1006 at public.ip.removed SOFIA DESTROY
> 2011-06-22 13:33:50.780056 [DEBUG] switch_nat.c:570 unmapped public port
> 31484 protocol UDP to localport 31484
> 2011-06-22 13:33:50.840070 [DEBUG] switch_nat.c:570 unmapped public port
> 31485 protocol UDP to localport 31485
> 2011-06-22 13:33:50.840070 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/1006 at public.ip.removed Standard DESTROY
> 2011-06-22 13:33:50.840070 [DEBUG] switch_core_state_machine.c:467
> (sofia/internal/1006 at public.ip.removed) State DESTROY going to sleep
>
> The above example was from an externally situated SIP phone ringing
> voicemail (4000) on FreeSWITCH.
>
> I have experimented changing various timers and timeouts in the config of
> FreeSWITCH (one at a time, being careful to put them back afterwards!) but
> been unable to resolve the issue.
>
> Incidentally, we have no long term intention of running off-site SIP phones
> with the PBX and I'm hoping not to have to leave it in the DMZ either, it's
> just like that for debugging. What is a real issue is the calls to our
> external SIP provider (i.e. outbound calls) being dropped.
>
> Any suggestions would be greatly appreciated.
>
> Thanks,
>
> Matthew Ralston
> Web Developer & IT Consultant
>
> matt at mralston.co.uk
> www.mralston.com
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sid.kshatriya at gmail.com  Wed Jun 22 17:39:09 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Wed, 22 Jun 2011 19:09:09 +0530
Subject: [Freeswitch-users] Looking for Cost Effective IVR Solution
 forConcurrent Calls
In-Reply-To: <7091CBE8A30F42978D9F082C267C22DA@sharad>
References: <BANLkTi=TLxCRTxkJ-B5nO92YVcGOpBtp0A@mail.gmail.com>
	<7091CBE8A30F42978D9F082C267C22DA@sharad>
Message-ID: <BANLkTimJfoz2PF7YHXHpWFaqkEHKgQ8y7Q@mail.gmail.com>

PRI pricing is about 4,500/- rupees a month. You can purchase a toll free
plan from a provider -- how much you use (or don't use) the toll free number
is upto you. I know this because I am in the process of purchasing a PRI.

On Wed, Jun 22, 2011 at 2:47 PM, sharad <sharad at coraltele.com> wrote:

> **
> Hi Ankit
>
> I differ with you on the PRI pricing. Now a days, in north India, BSNL,
> Reliance, Airtel, etc, are providing PRI connectivity somewhere arround
> 10-12 K per month. More over this, they will permit you to make the free
> calls of the same value. Means if you opt for 10K / month plan, you will be
> able to make the out calls of wrth Rs 10K approx. Yes, if you want only
> inbound, you will have to pay at least 10K.  So you can check with the local
> service provider in Tamilnadu & nagotiate with the service provider for
> minimum rent.
>
> Regarding your rest queries, I am not sure what do you want to ask..plz
> explore your queries.
>
> Regards
> Sharad, Noida
> 9891499202
>
>
>
> ----- Original Message -----
> *From:* ankIT WALiA <ankitwalia4u at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> ; The
> Linux-Delhi mailing list <ilugd at lists.linux-delhi.org> ;
> ilugc at ae.iitm.ac.in
> *Sent:* Wednesday, June 22, 2011 10:46 AM
> *Subject:* [Freeswitch-users] Looking for Cost Effective IVR Solution
> forConcurrent Calls
>
> Dear all,
>
> I am working to develop an IVR solution for an NGO in Tamil Nadu, India
> using Freeswitch open source telephony interface.
>
> I am looking for any cost effective solution for hardware where concurrent
> inbound/outbound calls can be made by the IVR system. Some research, I did
> but I am skeptical, need some more guidance from experts. Number of
> concurrent calls for initial stage inbound 5-10 but based on response from
> the community, we will plan for scaling up. Outbound Scheduled calls are of
> lesser importance right now.
>
> I checked online that PRI line provides me 30 lines for voice which can be
> used concurrently. But, the rent for the line is very high which cant be
> feasible for the NGO.
> Min PRI line cost is Rs 4.5 K per month.
>
> Please suggest alternative solutions which I can use for the same.
>
> 1. What about VOIP  and using their local access number?
> 2. What about SIP based IP phone with extensions?
> 3. What about trunk lines, DID etc?
> 4. What about GSM Modems, GSMOpen?
>
> I am very new telecom technology. Please help.
>
> Thanks
> Ankit
>
> --
> Life is like a rose its upto u feel it as its fragrance or thorns
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sidharth Kshatriya
www.sidk.info
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From henk at oegema.com  Wed Jun 22 18:03:38 2011
From: henk at oegema.com (Henk Oegema)
Date: Wed, 22 Jun 2011 16:03:38 +0200
Subject: [Freeswitch-users] FreeSWITCH-users@lists.freeswitch.org
Message-ID: <1308751418.2780.11.camel@DELL>

How come I don't see my mails in the FreeSWITCH-Users Archives?


Rgds.
Henk
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From wes-fs at 499x.com  Wed Jun 22 18:47:39 2011
From: wes-fs at 499x.com (Wes)
Date: Wed, 22 Jun 2011 09:47:39 -0500
Subject: [Freeswitch-users] Need help detecting when phone call is answered
 in lua script (it is starting before the call is picked up)
Message-ID: <4E02008B.7080908@499x.com>

Hello,

I'm having trouble detecting when a phone call (originated from 
freeswitch) is answered by the called party using a lua script.  I have 
freeswitch connected to a corporate phone system at my office via SIP.  
When I call my deskphone at work (914145551212), the script waits until 
I answer to start playing the welcome message, and other actions.  But 
when I call my cellphone (912625551212), the script starts before I pick 
up the cellphone, as can be seen in the logs I've pasted below.

Even though the lua script isn't properly detecting when the cell phone 
has been answered, freeswitch *does* seem to know, as shown by the line:
2011-06-22 09:21:28.334618 [NOTICE] sofia.c:5594 Channel 
[sofia/external/912625551212] has been answered
which gets logged when I answer the cell, even though the script is 
already running.

This fact gives me hope that there will be a way to fix this script.

Any suggestions are appreciated!
thanks.

using:
FreeSWITCH Version 1.0.head (git-8decee3 2011-06-20 13-21-20 -0500)

Script
http://pastebin.freeswitch.org/16564

logs:
http://pastebin.freeswitch.org/16563



From freeswitch at mralston.com  Wed Jun 22 18:49:51 2011
From: freeswitch at mralston.com (Matthew Ralston)
Date: Wed, 22 Jun 2011 15:49:51 +0100
Subject: [Freeswitch-users] Call drops after 30 seconds
In-Reply-To: <BANLkTikoP+hvtbzDn6fm_O6MmSN--x7TyA@mail.gmail.com>
References: <71F1700A-7641-43E5-A7C6-F25A0DC2EBC5@mralston.com>
	<BANLkTikoP+hvtbzDn6fm_O6MmSN--x7TyA@mail.gmail.com>
Message-ID: <C3BE3091-1685-42D1-ACD1-223B370607B7@mralston.com>

Hi Chris,

Thanks for the quick reply!

The FreeSWITCH box is in the DMZ of a Cisco Linksys WAG160N. Best I can tell, the DMZ is doing its job and allowing all ports through, inbound and outbound, TCP & UDP.

The external SIP phones are a Cisco SPA504G and Bria on the iPhone. These are behind behind a Cisco ASA5505, which has policy inspection for SIP switched on, i.e. ALG. I have also tested with Bria going over 3G (so it's not behind the Cisco ASA) and had the same problem.

The other scenario we have is some Yealink T20P SIP phones on the same LAN segment as the FreeSWITCH box. These can make A-leg only calls into FreeSWITCH (like calling voicemail) and also calls to other internal SIP phones fine. However when they make an outbound call the problem happens again. In this case the b-leg of the calls are sent to an external SIP provider and get cut after 30 seconds. Incidentally, we also use the same SIP provider from an Asterisk box in our data centre and that doesn't have a problem, so I believe the SIP provider is fine.

Kind regards,

Matthew Ralston
Web Developer & IT Consultant

matt at mralston.co.uk
www.mralston.com

On 22 Jun 2011, at 14:21, Chris Chen wrote:

> Hi Matthew, this is typical behavior for the setup of SIP behind NAT.
>    
>     1) Please provide the exact setup of remote SIP phones, what's the router model, does it have SIP ALG enabled, what kind  of SIP phones 
>     2)
> 
> 
> On Wed, Jun 22, 2011 at 8:38 AM, Matthew Ralston <freeswitch at mralston.com> wrote:
> Hi,
> 
> I'm having a problem at the moment with calls being successfully set up, with two-way audio, being terminated by FreeSWITCH after 30 seconds.
> 
> Internal calls (i.e. between SIP phones on the same LAN segment as the FreeSWITCH box) work flawlessly.
> 
> The problem arises when at least one of the handsets is located elsewhere on the Internet. This behaviour is exhibited under the following circumstances:
> 
> - A-leg only call, e.g. to voicemail when the handset is at another location on the Internet
> - A-leg-B-leg call if one or both of the handsets are at another location on the Internet
> - Inbound calls from our external SIP provider
> - Outbound calls to our external SIP provider
> 
> So it is obvious that the problem is related to the SIP going via the Internet, but I'm having trouble understanding why.
> 
> Whilst debugging this problem I have placed the FreeSWITCH box is in the DMZ on our router, so there should not be any ports blocked. The FreeSWITCH box itself is not running a software firewall.
> 
> The calls themselves are absolutely fine for the first 30 seconds - each party can hear the other talking fine.
> 
> The fact that the call is consistently dropped after 30 seconds (give or take a second or two for PDD) suggests that some timeout is being triggered.
> 
> When FreeSWITCH terminates the call, the following is logged to the console:
> 
> 2011-06-22 13:33:50.514941 [DEBUG] sofia.c:4787 Channel sofia/internal/1006 at public.ip.removed entering state [terminating][0]
> 2011-06-22 13:33:50.514941 [DEBUG] switch_channel.c:2641 (sofia/internal/1006 at public.ip.removed) Callstate Change ACTIVE -> HANGUP
> 2011-06-22 13:33:50.514941 [NOTICE] sofia.c:5508 Hangup sofia/internal/1006 at public.ip.removed [CS_EXECUTE] [NORMAL_UNSPECIFIED]
> 2011-06-22 13:33:50.514941 [DEBUG] switch_channel.c:2657 Send signal sofia/internal/1006 at public.ip.removed [KILL]
> 2011-06-22 13:33:50.514941 [DEBUG] switch_core_session.c:1118 Send signal sofia/internal/1006 at public.ip.removed [BREAK]
> 2011-06-22 13:33:50.534966 [DEBUG] switch_ivr_play_say.c:1649 done playing file
> 2011-06-22 13:33:50.625988 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-press.wav] (en:en)
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_session.c:2063 sofia/internal/1006 at public.ip.removed skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:371 (sofia/internal/1006 at public.ip.removed) State EXECUTE going to sleep
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1006 at public.ip.removed) Running State Change CS_HANGUP
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/1006 at public.ip.removed) State HANGUP
> 2011-06-22 13:33:50.727027 [DEBUG] mod_sofia.c:458 Channel sofia/internal/1006 at public.ip.removed hanging up, cause: NORMAL_UNSPECIFIED
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1006 at public.ip.removed Standard HANGUP, cause: NORMAL_UNSPECIFIED
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/1006 at public.ip.removed) State HANGUP going to sleep
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/1006 at public.ip.removed) State Change CS_HANGUP -> CS_REPORTING
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_session.c:1118 Send signal sofia/internal/1006 at public.ip.removed [BREAK]
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1006 at public.ip.removed) Running State Change CS_REPORTING
> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/1006 at public.ip.removed) State REPORTING
> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1006 at public.ip.removed Standard REPORTING, cause: NORMAL_UNSPECIFIED
> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/1006 at public.ip.removed) State REPORTING going to sleep
> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:350 (sofia/internal/1006 at public.ip.removed) State Change CS_REPORTING -> CS_DESTROY
> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_session.c:1118 Send signal sofia/internal/1006 at public.ip.removed [BREAK]
> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_session.c:1290 Session 5 (sofia/internal/1006 at public.ip.removed) Locked, Waiting on external entities
> 2011-06-22 13:33:50.740064 [NOTICE] switch_core_session.c:1308 Session 5 (sofia/internal/1006 at public.ip.removed) Ended
> 2011-06-22 13:33:50.740064 [NOTICE] switch_core_session.c:1310 Close Channel sofia/internal/1006 at public.ip.removed [CS_DESTROY]
> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/1006 at public.ip.removed) Callstate Change HANGUP -> DOWN
> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:457 (sofia/internal/1006 at public.ip.removed) Running State Change CS_DESTROY
> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1006 at public.ip.removed) State DESTROY
> 2011-06-22 13:33:50.740064 [DEBUG] mod_sofia.c:363 sofia/internal/1006 at public.ip.removed SOFIA DESTROY
> 2011-06-22 13:33:50.780056 [DEBUG] switch_nat.c:570 unmapped public port 31484 protocol UDP to localport 31484
> 2011-06-22 13:33:50.840070 [DEBUG] switch_nat.c:570 unmapped public port 31485 protocol UDP to localport 31485
> 2011-06-22 13:33:50.840070 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1006 at public.ip.removed Standard DESTROY
> 2011-06-22 13:33:50.840070 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1006 at public.ip.removed) State DESTROY going to sleep
> 
> The above example was from an externally situated SIP phone ringing voicemail (4000) on FreeSWITCH.
> 
> I have experimented changing various timers and timeouts in the config of FreeSWITCH (one at a time, being careful to put them back afterwards!) but been unable to resolve the issue.
> 
> Incidentally, we have no long term intention of running off-site SIP phones with the PBX and I'm hoping not to have to leave it in the DMZ either, it's just like that for debugging. What is a real issue is the calls to our external SIP provider (i.e. outbound calls) being dropped.
> 
> Any suggestions would be greatly appreciated.
> 
> Thanks,
> 
> Matthew Ralston
> Web Developer & IT Consultant
> 
> matt at mralston.co.uk
> www.mralston.com
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From infos at madovsky.org  Wed Jun 22 18:52:12 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 22 Jun 2011 10:52:12 -0400
Subject: [Freeswitch-users] mod_rtmp client API
Message-ID: <CFF7DAB2B60A478288D8B9919C7300D0@e1705>

Is there any client side API to use mod_rtmp ?

Thanks
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From max.asterisk at gmail.com  Wed Jun 22 14:23:35 2011
From: max.asterisk at gmail.com (Max Alex)
Date: Wed, 22 Jun 2011 15:53:35 +0530
Subject: [Freeswitch-users] Answer issue on inbound call
Message-ID: <BANLkTin+f=uDMiOouyLWAorPvNTd1Z6xaA@mail.gmail.com>

Hi,
I have installed freeswitch latest version with sangoma card A200 as well,
I have installed and configured freetdm module with wanpipe drivers for
freeswitch,
We are properly receiving the inbound calls in public context and then we
are routing that call to 1001 extension,
it is properly routing to 1001 as well, but we have one issue while routing
on 1001.

Here is the issue description.
I am calling from my cell phone to that DID number of pri line, and then it
will start ringing on 1001 extension,
When 1001 extension start ringing the call is answered on my cell phone,
it is something like freeswitch preanswer or autoanswer the call, how can i
stop this answer call when it is ringing on 1001 extension,
Waiting for good reply.

Thanks,
Max Alex
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From Hector.Geraldino at ip-soft.net  Wed Jun 22 18:54:41 2011
From: Hector.Geraldino at ip-soft.net (Hector Geraldino)
Date: Wed, 22 Jun 2011 10:54:41 -0400
Subject: [Freeswitch-users] FS and ASR engine
Message-ID: <6A6B4C284AD15042B429EB9D904544AD021F939AE4@NY1-EXMB-01.ip-soft.net>

Hi everyone,

I want to check with you guys to see if anyone has experience integrating FreeSwitch with an ASR engine, and using the engine as a merely transcriber of the conversation.

I've been playing for the past two weeks or so with Nuance Speech Server/recognizer and pocketsphinx. Nuance is by far a better solution, but due to the lack of freely available documentation and my short expertise in this subject, I haven't been to achieve my goal.

The communication between FS and the ASR engine works great using MCRP, my concern is with the ASR engine itself. I want to allow the user to speak freely, and get a transcription of what the user said. I don't want or need to understand what the meaning of the utterances are (definitely the engine doesn't need to do that), also I don't need/want to write any complex grammar or SLM to get an interpretation of the spoken phrases, I just want the plain text of what has been said. No decisions will be taken based on what the user said, this information will just be passed to a 3rd application.

I don't know if this can be achieved or not without developing grammars (not suitable for open-ended dialogs) or training statistical language models. What I do recall is using Dragon Speak in MS word for dictation, without the need of doing some trtraining or developing grammars. That's exactly what I'm pursuing: a simple plain text transcription of the spoken words.

Have anyone of you deal with something like this by any chance?

Thanks for your help. I apologize if this is not the right place to ask this type of questions.

Thanks again,
Hector
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From msc at freeswitch.org  Wed Jun 22 19:11:14 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 22 Jun 2011 08:11:14 -0700
Subject: [Freeswitch-users] mod_rtmp client API
In-Reply-To: <CFF7DAB2B60A478288D8B9919C7300D0@e1705>
References: <CFF7DAB2B60A478288D8B9919C7300D0@e1705>
Message-ID: <BANLkTimS4GbPF99_K-G9iufjSbmAWDK=fg@mail.gmail.com>

No, it's a DIY project. We provided the server, you provide the client. :D
Google around and you will find various tools and pieces to start building
your own.

-MC

On Wed, Jun 22, 2011 at 7:52 AM, Madovsky <infos at madovsky.org> wrote:

> **
> Is there any client side API to use mod_rtmp ?
>
> Thanks
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chris.chen2004 at gmail.com  Wed Jun 22 19:11:13 2011
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Wed, 22 Jun 2011 11:11:13 -0400
Subject: [Freeswitch-users] Call drops after 30 seconds
In-Reply-To: <C3BE3091-1685-42D1-ACD1-223B370607B7@mralston.com>
References: <71F1700A-7641-43E5-A7C6-F25A0DC2EBC5@mralston.com>
	<BANLkTikoP+hvtbzDn6fm_O6MmSN--x7TyA@mail.gmail.com>
	<C3BE3091-1685-42D1-ACD1-223B370607B7@mralston.com>
Message-ID: <BANLkTi=4e1t607G0xCRygHGotxQrsXcnzw@mail.gmail.com>

Hi Matthew, continue my last reply here

2) Is your FS server using private IP address?  you have to setup your FS
external SIP/RTP IP address to the proper public IP address, by either using
UPNP enabled router, STUN, or hardcoded public IP address in sofia profiles.

Please check that.

Thanks,
Chris


On Wed, Jun 22, 2011 at 10:49 AM, Matthew Ralston
<freeswitch at mralston.com>wrote:

> Hi Chris,
>
> Thanks for the quick reply!
>
> The FreeSWITCH box is in the DMZ of a Cisco Linksys WAG160N. Best I can
> tell, the DMZ is doing its job and allowing all ports through, inbound and
> outbound, TCP & UDP.
>
> The external SIP phones are a Cisco SPA504G and Bria on the iPhone. These
> are behind behind a Cisco ASA5505, which has policy inspection for SIP
> switched on, i.e. ALG. I have also tested with Bria going over 3G (so it's
> not behind the Cisco ASA) and had the same problem.
>
> The other scenario we have is some Yealink T20P SIP phones on the same LAN
> segment as the FreeSWITCH box. These can make A-leg only calls into
> FreeSWITCH (like calling voicemail) and also calls to other internal SIP
> phones fine. However when they make an outbound call the problem happens
> again. In this case the b-leg of the calls are sent to an external SIP
> provider and get cut after 30 seconds. Incidentally, we also use the same
> SIP provider from an Asterisk box in our data centre and that doesn't have a
> problem, so I believe the SIP provider is fine.
>
> Kind regards,
>
>
> Matthew Ralston
> Web Developer & IT Consultant
>
> matt at mralston.co.uk
> www.mralston.com
>
> On 22 Jun 2011, at 14:21, Chris Chen wrote:
>
> Hi Matthew, this is typical behavior for the setup of SIP behind NAT.
>
>     1) Please provide the exact setup of remote SIP phones, what's the
> router model, does it have SIP ALG enabled, what kind  of SIP phones
>     2)
>
>
> On Wed, Jun 22, 2011 at 8:38 AM, Matthew Ralston <freeswitch at mralston.com>wrote:
>
>> Hi,
>>
>> I'm having a problem at the moment with calls being successfully set up,
>> with two-way audio, being terminated by FreeSWITCH after 30 seconds.
>>
>> Internal calls (i.e. between SIP phones on the same LAN segment as the
>> FreeSWITCH box) work flawlessly.
>>
>> The problem arises when at least one of the handsets is located elsewhere
>> on the Internet. This behaviour is exhibited under the following
>> circumstances:
>>
>> - A-leg only call, e.g. to voicemail when the handset is at another
>> location on the Internet
>> - A-leg-B-leg call if one or both of the handsets are at another location
>> on the Internet
>> - Inbound calls from our external SIP provider
>> - Outbound calls to our external SIP provider
>>
>> So it is obvious that the problem is related to the SIP going via the
>> Internet, but I'm having trouble understanding why.
>>
>> Whilst debugging this problem I have placed the FreeSWITCH box is in the
>> DMZ on our router, so there should not be any ports blocked. The FreeSWITCH
>> box itself is not running a software firewall.
>>
>> The calls themselves are absolutely fine for the first 30 seconds - each
>> party can hear the other talking fine.
>>
>> The fact that the call is consistently dropped after 30 seconds (give or
>> take a second or two for PDD) suggests that some timeout is being triggered.
>>
>> When FreeSWITCH terminates the call, the following is logged to the
>> console:
>>
>> 2011-06-22 13:33:50.514941 [DEBUG] sofia.c:4787 Channel
>> sofia/internal/1006 at public.ip.removed entering state [terminating][0]
>> 2011-06-22 13:33:50.514941 [DEBUG] switch_channel.c:2641 (
>> sofia/internal/1006 at public.ip.removed) Callstate Change ACTIVE -> HANGUP
>> 2011-06-22 13:33:50.514941 [NOTICE] sofia.c:5508 Hangup
>> sofia/internal/1006 at public.ip.removed [CS_EXECUTE] [NORMAL_UNSPECIFIED]
>> 2011-06-22 13:33:50.514941 [DEBUG] switch_channel.c:2657 Send signal
>> sofia/internal/1006 at public.ip.removed [KILL]
>> 2011-06-22 13:33:50.514941 [DEBUG] switch_core_session.c:1118 Send signal
>> sofia/internal/1006 at public.ip.removed [BREAK]
>> 2011-06-22 13:33:50.534966 [DEBUG] switch_ivr_play_say.c:1649 done playing
>> file
>> 2011-06-22 13:33:50.625988 [DEBUG] switch_ivr_play_say.c:244 Handle
>> play-file:[voicemail/vm-press.wav] (en:en)
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_session.c:2063
>> sofia/internal/1006 at public.ip.removed skip receive message
>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:371 (
>> sofia/internal/1006 at public.ip.removed) State EXECUTE going to sleep
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:325 (
>> sofia/internal/1006 at public.ip.removed) Running State Change CS_HANGUP
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:565 (
>> sofia/internal/1006 at public.ip.removed) State HANGUP
>> 2011-06-22 13:33:50.727027 [DEBUG] mod_sofia.c:458 Channel
>> sofia/internal/1006 at public.ip.removed hanging up, cause:
>> NORMAL_UNSPECIFIED
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:46
>> sofia/internal/1006 at public.ip.removed Standard HANGUP, cause:
>> NORMAL_UNSPECIFIED
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:565 (
>> sofia/internal/1006 at public.ip.removed) State HANGUP going to sleep
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:356 (
>> sofia/internal/1006 at public.ip.removed) State Change CS_HANGUP ->
>> CS_REPORTING
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_session.c:1118 Send signal
>> sofia/internal/1006 at public.ip.removed [BREAK]
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:325 (
>> sofia/internal/1006 at public.ip.removed) Running State Change CS_REPORTING
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:625 (
>> sofia/internal/1006 at public.ip.removed) State REPORTING
>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:53
>> sofia/internal/1006 at public.ip.removed Standard REPORTING, cause:
>> NORMAL_UNSPECIFIED
>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:625 (
>> sofia/internal/1006 at public.ip.removed) State REPORTING going to sleep
>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:350 (
>> sofia/internal/1006 at public.ip.removed) State Change CS_REPORTING ->
>> CS_DESTROY
>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_session.c:1118 Send signal
>> sofia/internal/1006 at public.ip.removed [BREAK]
>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_session.c:1290 Session 5 (
>> sofia/internal/1006 at public.ip.removed) Locked, Waiting on external
>> entities
>> 2011-06-22 13:33:50.740064 [NOTICE] switch_core_session.c:1308 Session 5 (
>> sofia/internal/1006 at public.ip.removed) Ended
>> 2011-06-22 13:33:50.740064 [NOTICE] switch_core_session.c:1310 Close
>> Channel sofia/internal/1006 at public.ip.removed [CS_DESTROY]
>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:454 (
>> sofia/internal/1006 at public.ip.removed) Callstate Change HANGUP -> DOWN
>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:457 (
>> sofia/internal/1006 at public.ip.removed) Running State Change CS_DESTROY
>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:467 (
>> sofia/internal/1006 at public.ip.removed) State DESTROY
>> 2011-06-22 13:33:50.740064 [DEBUG] mod_sofia.c:363
>> sofia/internal/1006 at public.ip.removed SOFIA DESTROY
>> 2011-06-22 13:33:50.780056 [DEBUG] switch_nat.c:570 unmapped public port
>> 31484 protocol UDP to localport 31484
>> 2011-06-22 13:33:50.840070 [DEBUG] switch_nat.c:570 unmapped public port
>> 31485 protocol UDP to localport 31485
>> 2011-06-22 13:33:50.840070 [DEBUG] switch_core_state_machine.c:60
>> sofia/internal/1006 at public.ip.removed Standard DESTROY
>> 2011-06-22 13:33:50.840070 [DEBUG] switch_core_state_machine.c:467 (
>> sofia/internal/1006 at public.ip.removed) State DESTROY going to sleep
>>
>> The above example was from an externally situated SIP phone ringing
>> voicemail (4000) on FreeSWITCH.
>>
>> I have experimented changing various timers and timeouts in the config of
>> FreeSWITCH (one at a time, being careful to put them back afterwards!) but
>> been unable to resolve the issue.
>>
>> Incidentally, we have no long term intention of running off-site SIP
>> phones with the PBX and I'm hoping not to have to leave it in the DMZ
>> either, it's just like that for debugging. What is a real issue is the calls
>> to our external SIP provider (i.e. outbound calls) being dropped.
>>
>> Any suggestions would be greatly appreciated.
>>
>> Thanks,
>>
>> Matthew Ralston
>> Web Developer & IT Consultant
>>
>> matt at mralston.co.uk
>> www.mralston.com
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Wed Jun 22 19:20:24 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 22 Jun 2011 11:20:24 -0400
Subject: [Freeswitch-users] mod_rtmp client API
References: <CFF7DAB2B60A478288D8B9919C7300D0@e1705>
	<BANLkTimS4GbPF99_K-G9iufjSbmAWDK=fg@mail.gmail.com>
Message-ID: <F4E8D1CEA93240CBA52E72CECFAC8B0F@e1705>

Mike,

I didn't ask tools, but to know at least which function it needs to call to the server
to make it work ! :)
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, June 22, 2011 11:11 AM
  Subject: Re: [Freeswitch-users] mod_rtmp client API


  No, it's a DIY project. We provided the server, you provide the client. :D Google around and you will find various tools and pieces to start building your own.


  -MC


  On Wed, Jun 22, 2011 at 7:52 AM, Madovsky <infos at madovsky.org> wrote:

    Is there any client side API to use mod_rtmp ?

    Thanks

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

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






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


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

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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From msc at freeswitch.org  Wed Jun 22 19:21:22 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 22 Jun 2011 08:21:22 -0700
Subject: [Freeswitch-users] Answer issue on inbound call
In-Reply-To: <BANLkTin+f=uDMiOouyLWAorPvNTd1Z6xaA@mail.gmail.com>
References: <BANLkTin+f=uDMiOouyLWAorPvNTd1Z6xaA@mail.gmail.com>
Message-ID: <BANLkTim=TOQD1tUvmHjeRuhAR-eKkGYgsw@mail.gmail.com>

I thought the A200 was an analog card? Maybe I have my numbers mixed up...

Go ahead and collect a debug log of this call. It might help to have your
configs posted as well. Use pastebin.freeswitch.org. See this wiki article
for tips on how to collect information:
http://wiki.freeswitch.org/wiki/Reporting_Bugs

-MC

On Wed, Jun 22, 2011 at 3:23 AM, Max Alex <max.asterisk at gmail.com> wrote:

> Hi,
> I have installed freeswitch latest version with sangoma card A200 as well,
> I have installed and configured freetdm module with wanpipe drivers for
> freeswitch,
> We are properly receiving the inbound calls in public context and then we
> are routing that call to 1001 extension,
> it is properly routing to 1001 as well, but we have one issue while routing
> on 1001.
>
> Here is the issue description.
> I am calling from my cell phone to that DID number of pri line, and then it
> will start ringing on 1001 extension,
> When 1001 extension start ringing the call is answered on my cell phone,
> it is something like freeswitch preanswer or autoanswer the call, how can i
> stop this answer call when it is ringing on 1001 extension,
> Waiting for good reply.
>
> Thanks,
> Max Alex
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
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From freeswitch at mralston.com  Wed Jun 22 19:26:43 2011
From: freeswitch at mralston.com (Matthew Ralston)
Date: Wed, 22 Jun 2011 16:26:43 +0100
Subject: [Freeswitch-users] Call drops after 30 seconds
In-Reply-To: <BANLkTi=4e1t607G0xCRygHGotxQrsXcnzw@mail.gmail.com>
References: <71F1700A-7641-43E5-A7C6-F25A0DC2EBC5@mralston.com>
	<BANLkTikoP+hvtbzDn6fm_O6MmSN--x7TyA@mail.gmail.com>
	<C3BE3091-1685-42D1-ACD1-223B370607B7@mralston.com>
	<BANLkTi=4e1t607G0xCRygHGotxQrsXcnzw@mail.gmail.com>
Message-ID: <28830F22-9654-45D2-B00A-1E69E32A42F9@mralston.com>

Hi Chris,

The FreeSWITCH box does indeed have a private IP address. The Cisco Linksys WAG160N router in front of it has UPnP enabled, not that I particularly trust it.

So maybe it's worth hard coding the public IP into the Sofia config.

Where is the best place to put the public IP? The internal.xml and external.xml files both refer to $${local_ip_v4} although I don't see where this comes from. Does FreeSWITCH figure this out itself in some way?

As I have internal SIP phones, an external SIP provider and (whilst debugging) some external SIP phones (which use the internal profile!!). I'm concerned that if I hard code the public IP address in to the wrong place it will cause problems for the internal SIP phones.

Kind regards,

Matthew Ralston
Web Developer & IT Consultant

matt at mralston.co.uk
www.mralston.com

On 22 Jun 2011, at 16:11, Chris Chen wrote:

> Hi Matthew, continue my last reply here
> 
> 2) Is your FS server using private IP address?  you have to setup your FS external SIP/RTP IP address to the proper public IP address, by either using UPNP enabled router, STUN, or hardcoded public IP address in sofia profiles.
> 
> Please check that.
> 
> Thanks,
> Chris 
>  
> 
> On Wed, Jun 22, 2011 at 10:49 AM, Matthew Ralston <freeswitch at mralston.com> wrote:
> Hi Chris,
> 
> Thanks for the quick reply!
> 
> The FreeSWITCH box is in the DMZ of a Cisco Linksys WAG160N. Best I can tell, the DMZ is doing its job and allowing all ports through, inbound and outbound, TCP & UDP.
> 
> The external SIP phones are a Cisco SPA504G and Bria on the iPhone. These are behind behind a Cisco ASA5505, which has policy inspection for SIP switched on, i.e. ALG. I have also tested with Bria going over 3G (so it's not behind the Cisco ASA) and had the same problem.
> 
> The other scenario we have is some Yealink T20P SIP phones on the same LAN segment as the FreeSWITCH box. These can make A-leg only calls into FreeSWITCH (like calling voicemail) and also calls to other internal SIP phones fine. However when they make an outbound call the problem happens again. In this case the b-leg of the calls are sent to an external SIP provider and get cut after 30 seconds. Incidentally, we also use the same SIP provider from an Asterisk box in our data centre and that doesn't have a problem, so I believe the SIP provider is fine.
> 
> Kind regards,
> 
> 
> Matthew Ralston
> Web Developer & IT Consultant
> 
> matt at mralston.co.uk
> www.mralston.com
> 
> On 22 Jun 2011, at 14:21, Chris Chen wrote:
> 
>> Hi Matthew, this is typical behavior for the setup of SIP behind NAT.
>>    
>>     1) Please provide the exact setup of remote SIP phones, what's the router model, does it have SIP ALG enabled, what kind  of SIP phones 
>>     2)
>> 
>> 
>> On Wed, Jun 22, 2011 at 8:38 AM, Matthew Ralston <freeswitch at mralston.com> wrote:
>> Hi,
>> 
>> I'm having a problem at the moment with calls being successfully set up, with two-way audio, being terminated by FreeSWITCH after 30 seconds.
>> 
>> Internal calls (i.e. between SIP phones on the same LAN segment as the FreeSWITCH box) work flawlessly.
>> 
>> The problem arises when at least one of the handsets is located elsewhere on the Internet. This behaviour is exhibited under the following circumstances:
>> 
>> - A-leg only call, e.g. to voicemail when the handset is at another location on the Internet
>> - A-leg-B-leg call if one or both of the handsets are at another location on the Internet
>> - Inbound calls from our external SIP provider
>> - Outbound calls to our external SIP provider
>> 
>> So it is obvious that the problem is related to the SIP going via the Internet, but I'm having trouble understanding why.
>> 
>> Whilst debugging this problem I have placed the FreeSWITCH box is in the DMZ on our router, so there should not be any ports blocked. The FreeSWITCH box itself is not running a software firewall.
>> 
>> The calls themselves are absolutely fine for the first 30 seconds - each party can hear the other talking fine.
>> 
>> The fact that the call is consistently dropped after 30 seconds (give or take a second or two for PDD) suggests that some timeout is being triggered.
>> 
>> When FreeSWITCH terminates the call, the following is logged to the console:
>> 
>> 2011-06-22 13:33:50.514941 [DEBUG] sofia.c:4787 Channel sofia/internal/1006 at public.ip.removed entering state [terminating][0]
>> 2011-06-22 13:33:50.514941 [DEBUG] switch_channel.c:2641 (sofia/internal/1006 at public.ip.removed) Callstate Change ACTIVE -> HANGUP
>> 2011-06-22 13:33:50.514941 [NOTICE] sofia.c:5508 Hangup sofia/internal/1006 at public.ip.removed [CS_EXECUTE] [NORMAL_UNSPECIFIED]
>> 2011-06-22 13:33:50.514941 [DEBUG] switch_channel.c:2657 Send signal sofia/internal/1006 at public.ip.removed [KILL]
>> 2011-06-22 13:33:50.514941 [DEBUG] switch_core_session.c:1118 Send signal sofia/internal/1006 at public.ip.removed [BREAK]
>> 2011-06-22 13:33:50.534966 [DEBUG] switch_ivr_play_say.c:1649 done playing file
>> 2011-06-22 13:33:50.625988 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-press.wav] (en:en)
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_session.c:2063 sofia/internal/1006 at public.ip.removed skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:371 (sofia/internal/1006 at public.ip.removed) State EXECUTE going to sleep
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1006 at public.ip.removed) Running State Change CS_HANGUP
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/1006 at public.ip.removed) State HANGUP
>> 2011-06-22 13:33:50.727027 [DEBUG] mod_sofia.c:458 Channel sofia/internal/1006 at public.ip.removed hanging up, cause: NORMAL_UNSPECIFIED
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1006 at public.ip.removed Standard HANGUP, cause: NORMAL_UNSPECIFIED
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/1006 at public.ip.removed) State HANGUP going to sleep
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/1006 at public.ip.removed) State Change CS_HANGUP -> CS_REPORTING
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_session.c:1118 Send signal sofia/internal/1006 at public.ip.removed [BREAK]
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1006 at public.ip.removed) Running State Change CS_REPORTING
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/1006 at public.ip.removed) State REPORTING
>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1006 at public.ip.removed Standard REPORTING, cause: NORMAL_UNSPECIFIED
>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/1006 at public.ip.removed) State REPORTING going to sleep
>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:350 (sofia/internal/1006 at public.ip.removed) State Change CS_REPORTING -> CS_DESTROY
>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_session.c:1118 Send signal sofia/internal/1006 at public.ip.removed [BREAK]
>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_session.c:1290 Session 5 (sofia/internal/1006 at public.ip.removed) Locked, Waiting on external entities
>> 2011-06-22 13:33:50.740064 [NOTICE] switch_core_session.c:1308 Session 5 (sofia/internal/1006 at public.ip.removed) Ended
>> 2011-06-22 13:33:50.740064 [NOTICE] switch_core_session.c:1310 Close Channel sofia/internal/1006 at public.ip.removed [CS_DESTROY]
>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/1006 at public.ip.removed) Callstate Change HANGUP -> DOWN
>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:457 (sofia/internal/1006 at public.ip.removed) Running State Change CS_DESTROY
>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1006 at public.ip.removed) State DESTROY
>> 2011-06-22 13:33:50.740064 [DEBUG] mod_sofia.c:363 sofia/internal/1006 at public.ip.removed SOFIA DESTROY
>> 2011-06-22 13:33:50.780056 [DEBUG] switch_nat.c:570 unmapped public port 31484 protocol UDP to localport 31484
>> 2011-06-22 13:33:50.840070 [DEBUG] switch_nat.c:570 unmapped public port 31485 protocol UDP to localport 31485
>> 2011-06-22 13:33:50.840070 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1006 at public.ip.removed Standard DESTROY
>> 2011-06-22 13:33:50.840070 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1006 at public.ip.removed) State DESTROY going to sleep
>> 
>> The above example was from an externally situated SIP phone ringing voicemail (4000) on FreeSWITCH.
>> 
>> I have experimented changing various timers and timeouts in the config of FreeSWITCH (one at a time, being careful to put them back afterwards!) but been unable to resolve the issue.
>> 
>> Incidentally, we have no long term intention of running off-site SIP phones with the PBX and I'm hoping not to have to leave it in the DMZ either, it's just like that for debugging. What is a real issue is the calls to our external SIP provider (i.e. outbound calls) being dropped.
>> 
>> Any suggestions would be greatly appreciated.
>> 
>> Thanks,
>> 
>> Matthew Ralston
>> Web Developer & IT Consultant
>> 
>> matt at mralston.co.uk
>> www.mralston.com
>> 
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From a.luppi at seletech.com  Wed Jun 22 19:28:17 2011
From: a.luppi at seletech.com (Alessandro)
Date: Wed, 22 Jun 2011 17:28:17 +0200
Subject: [Freeswitch-users] Modified Follow me
In-Reply-To: <BANLkTimSVMrTGJg72JJe5252z7J-RMXSLA@mail.gmail.com>
References: <4E00A61C.1060503@seletech.com>
	<BANLkTimSVMrTGJg72JJe5252z7J-RMXSLA@mail.gmail.com>
Message-ID: <4E020A11.4080706@seletech.com>

Hi,

this is the v_huntgroup_localhost_$EXTENSION.lua
http://pastebin.freeswitch.org/16565

if the second line is deleted one problem is solved, FS doesn't answer 
with 200 ok, but answer with ringing.
But I'm not able to solve the other problem...
/    if the first extension is reachable and the user hangup before 
answering (hangup in ringing), the follow me  procedure starts to 
contact the second extension in the list. I want         that FS     
start to contact the second phone only if the first extension isn't 
registered to FS. /

I think is related with the line 47.

Some one can help me?

Thanks

Alessandro Luppi



Il 22/06/2011 10:42, Christian Benke ha scritto:
> On 21 June 2011 16:09, Alessandro<a.luppi at seletech.com>  wrote:
>> Any suggestion?
> While i can't give you a concrete solution, take a look at the files
> that are created by fusionpbx, mainly
> /usr/local/freeswitch/scripts/v_huntgroup_localhost_$EXTENSION.lua -
> if you dig a bit you might be able to find and fix the problem
> yourself(And eventually file a bugreport with fusionpbx).
>
> hth
> Christian
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
FIRST MAIL:
/Hi,

i'd like to implement this particular behaviour. If I call an extension 
that isn't registered to FS, FS re-route the call to another 
destination. If also the second destination in unreachable the call end. 
I first used the follow me (created with fusionpbx) to implement this 
feature, but I have the following problem: if the first extension is 
reachable and the user hangup before answering (hangup in ringing), the 
follow me procedure starts to contact the second extension in the list. 
I want that FS start to contact the second phone only if the first 
extension isn't registered to FS.
The second problem is that with follow-me FS answer with 200 OK at the 
invite of the caller bypassing the ringing status, than the procedure of 
follow-me is transparent to the caller.
I'd like that the caller stay in ringing status until one of the 
destination extension answer at the call or the call is closed by FS.
Any suggestion?

Regards

Alessandro Luppi /


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

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From infos at madovsky.org  Wed Jun 22 19:53:05 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 22 Jun 2011 11:53:05 -0400
Subject: [Freeswitch-users] mod_rtmp client API
Message-ID: <801B6B74CA73484F96B9BBD9BE125561@e1705>

if I type "actionscript mod_rtmp freeswitch" on google
nothing that helps a common developer to work on mod_rtmp
....

  ----- Original Message ----- 
  From: Madovsky 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, June 22, 2011 11:20 AM
  Subject: Re: [Freeswitch-users] mod_rtmp client API


  Mike,

  I didn't ask tools, but to know at least which function it needs to call to the server
  to make it work ! :)
    ----- Original Message ----- 
    From: Michael Collins 
    To: FreeSWITCH Users Help 
    Sent: Wednesday, June 22, 2011 11:11 AM
    Subject: Re: [Freeswitch-users] mod_rtmp client API


    No, it's a DIY project. We provided the server, you provide the client. :D Google around and you will find various tools and pieces to start building your own. 


    -MC


    On Wed, Jun 22, 2011 at 7:52 AM, Madovsky <infos at madovsky.org> wrote:

      Is there any client side API to use mod_rtmp ?

      Thanks

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

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






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


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

    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From pehr at harqen.com  Wed Jun 22 20:16:47 2011
From: pehr at harqen.com (Pehr Anderson)
Date: Wed, 22 Jun 2011 11:16:47 -0500
Subject: [Freeswitch-users] FS and ASR engine
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD021F939AE4@NY1-EXMB-01.ip-soft.net>
References: <6A6B4C284AD15042B429EB9D904544AD021F939AE4@NY1-EXMB-01.ip-soft.net>
Message-ID: <BANLkTik=PR3HB0mmNpF+2LG2ybdhPsEp_A@mail.gmail.com>

You might check out http://Nexiwave.com
They have been active at Cluecon and have a web API
that does fully hosted ASR on WAV's or MP3's.

They are doing innovation around running their ASR in GPU clusters,
which means their internal cost of operation is likely to be the lowest in
the industry.

Good ASR is always going to be computationally intensive,
so it is helpful to have somebody managing that for you.
It's going to be a lot easier that trying to juggle your own sphinx training
sets.

http://nexiwave.com/index.php/pricing

  --pehr

On Wed, Jun 22, 2011 at 9:54 AM, Hector Geraldino <
Hector.Geraldino at ip-soft.net> wrote:

> Hi everyone,****
>
> ** **
>
> I want to check with you guys to see if anyone has experience integrating
> FreeSwitch with an ASR engine, and using the engine as a merely transcriber
> of the conversation. ****
>
> ** **
>
> I?ve been playing for the past two weeks or so with Nuance Speech
> Server/recognizer and pocketsphinx. Nuance is by far a better solution, but
> due to the lack of freely available documentation and my short expertise in
> this subject, I haven?t been to achieve my goal.****
>
> ** **
>
> The communication between FS and the ASR engine works great using MCRP, my
> concern is with the ASR engine itself. I want to allow the user to speak
> freely, and get a transcription of what the user said. I don?t want or need
> to understand what the meaning of the utterances are (definitely the engine
> doesn?t need to do that), also I don?t need/want to write any complex
> grammar or SLM to get an interpretation of the spoken phrases, I just want
> the plain text of what has been said. No decisions will be taken based on
> what the user said, this information will just be passed to a 3rdapplication.
> ****
>
> ** **
>
> I don?t know if this can be achieved or not without developing grammars
> (not suitable for open-ended dialogs) or training statistical language
> models. What I do recall is using Dragon Speak in MS word for dictation,
> without the need of doing some trtraining or developing grammars. That?s
> exactly what I?m pursuing: a simple plain text transcription of the spoken
> words.****
>
> ** **
>
> Have anyone of you deal with something like this by any chance?****
>
> ** **
>
> Thanks for your help. I apologize if this is not the right place to ask
> this type of questions.****
>
> ** **
>
> Thanks again,****
>
> Hector****
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Pehr Anderson, VP Platform Technology
HarQen - http://HarQen.com
414-755-1962 x114
pehr at harqen.com
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From anthony.minessale at gmail.com  Wed Jun 22 20:19:20 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 22 Jun 2011 11:19:20 -0500
Subject: [Freeswitch-users] mod_rtmp client API
In-Reply-To: <801B6B74CA73484F96B9BBD9BE125561@e1705>
References: <801B6B74CA73484F96B9BBD9BE125561@e1705>
Message-ID: <BANLkTikmOfC7ot12-sGcO3aZ7+bH_cyGCQ@mail.gmail.com>

Actually......

We have something we are working on that we are considering releasing
if we can build a community base effort around it.
When I get a sign of enough interest we'll go from there.

It would be nice to do some presentations at ClueCon about it.


On Wed, Jun 22, 2011 at 10:53 AM, Madovsky <infos at madovsky.org> wrote:
> if I type "actionscript mod_rtmp freeswitch" on google
> nothing that helps a common developer to work on mod_rtmp
> ....
>
>
> ----- Original Message -----
> From: Madovsky
> To: FreeSWITCH Users Help
> Sent: Wednesday, June 22, 2011 11:20 AM
> Subject: Re: [Freeswitch-users] mod_rtmp client API
> Mike,
>
> I didn't ask tools, but to know at least which function it needs to call to
> the server
> to make it work ! :)
>
> ----- Original Message -----
> From: Michael Collins
> To: FreeSWITCH Users Help
> Sent: Wednesday, June 22, 2011 11:11 AM
> Subject: Re: [Freeswitch-users] mod_rtmp client API
> No, it's a DIY project. We provided the server, you provide the client. :D
> Google around and you will find various tools and pieces to start building
> your own.
> -MC
>
> On Wed, Jun 22, 2011 at 7:52 AM, Madovsky <infos at madovsky.org> wrote:
>>
>> Is there any client side API to use mod_rtmp ?
>>
>> Thanks
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> ________________________________
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From msc at freeswitch.org  Wed Jun 22 20:23:34 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 22 Jun 2011 09:23:34 -0700
Subject: [Freeswitch-users] mod_rtmp client API
In-Reply-To: <BANLkTikmOfC7ot12-sGcO3aZ7+bH_cyGCQ@mail.gmail.com>
References: <801B6B74CA73484F96B9BBD9BE125561@e1705>
	<BANLkTikmOfC7ot12-sGcO3aZ7+bH_cyGCQ@mail.gmail.com>
Message-ID: <BANLkTi=avuCv=TLSWaphhHuunVH6p_EJNg@mail.gmail.com>

On Wed, Jun 22, 2011 at 9:19 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Actually......
>
> We have something we are working on that we are considering releasing
> if we can build a community base effort around it.
> When I get a sign of enough interest we'll go from there.
>
> It would be nice to do some presentations at ClueCon about it.
>

I stand corrected. Thanks for the update, Anthony.
-<C
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From msc at freeswitch.org  Wed Jun 22 20:29:41 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 22 Jun 2011 09:29:41 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <BANLkTinw_PnBvgBAZ0_ePhrjhWgy1hbjBA@mail.gmail.com>

Hey folks,
Our conf call agenda is here:
http://wiki.freeswitch.org/wiki/FS_weekly_2011_06_22

Please add your items. It turns out our speaker for today had to reschedule
at the last minute so we're going to do a community scrum. Bring your topics
for conversation!

Thanks,
MC
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From infos at madovsky.org  Wed Jun 22 20:33:34 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 22 Jun 2011 12:33:34 -0400
Subject: [Freeswitch-users] mod_rtmp client API
References: <801B6B74CA73484F96B9BBD9BE125561@e1705>
	<BANLkTikmOfC7ot12-sGcO3aZ7+bH_cyGCQ@mail.gmail.com>
Message-ID: <F7AC533261894AD2856214A550404626@e1705>

Ok Anth,

adobe is actually going into the voip market very quickly,
they started some voip services already.
so if FS want a chance to create a community for mod_rtmp
it needs at least a minimum information to use it, if not, I can't see
the interest of mod_rtmp

Regards

----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, June 22, 2011 12:19 PM
Subject: Re: [Freeswitch-users] mod_rtmp client API


> Actually......
>
> We have something we are working on that we are considering releasing
> if we can build a community base effort around it.
> When I get a sign of enough interest we'll go from there.
>
> It would be nice to do some presentations at ClueCon about it.
>
>
> On Wed, Jun 22, 2011 at 10:53 AM, Madovsky <infos at madovsky.org> wrote:
>> if I type "actionscript mod_rtmp freeswitch" on google
>> nothing that helps a common developer to work on mod_rtmp
>> ....
>>
>>
>> ----- Original Message -----
>> From: Madovsky
>> To: FreeSWITCH Users Help
>> Sent: Wednesday, June 22, 2011 11:20 AM
>> Subject: Re: [Freeswitch-users] mod_rtmp client API
>> Mike,
>>
>> I didn't ask tools, but to know at least which function it needs to call 
>> to
>> the server
>> to make it work ! :)
>>
>> ----- Original Message -----
>> From: Michael Collins
>> To: FreeSWITCH Users Help
>> Sent: Wednesday, June 22, 2011 11:11 AM
>> Subject: Re: [Freeswitch-users] mod_rtmp client API
>> No, it's a DIY project. We provided the server, you provide the client. 
>> :D
>> Google around and you will find various tools and pieces to start 
>> building
>> your own.
>> -MC
>>
>> On Wed, Jun 22, 2011 at 7:52 AM, Madovsky <infos at madovsky.org> wrote:
>>>
>>> Is there any client side API to use mod_rtmp ?
>>>
>>> Thanks
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> ________________________________
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From Hector.Geraldino at ip-soft.net  Wed Jun 22 20:43:53 2011
From: Hector.Geraldino at ip-soft.net (Hector Geraldino)
Date: Wed, 22 Jun 2011 12:43:53 -0400
Subject: [Freeswitch-users] FS and ASR engine
In-Reply-To: <BANLkTik=PR3HB0mmNpF+2LG2ybdhPsEp_A@mail.gmail.com>
References: <6A6B4C284AD15042B429EB9D904544AD021F939AE4@NY1-EXMB-01.ip-soft.net>
	<BANLkTik=PR3HB0mmNpF+2LG2ybdhPsEp_A@mail.gmail.com>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD021F939AFD@NY1-EXMB-01.ip-soft.net>

Yeah, well, the thing is that it needs to be interactive (like an IVR system), so recording the voice and using a 3rd party service to translate a file  is not an option for me right now. That's why I'm using MCRP to communicate FreeSwitch with the ASR engine.

All I need is to figure out how to get the recognized text from a given ASR, without constraining the user to talk in an specific way (by a grammar) and, if possible, without training SLMs. My guess is that it's not possible, as I can't find any resources on the web showing this feature, but I'm still optimistic hoping to find an 'easy' way of doing the transcription.

From: Pehr Anderson [mailto:pehr at harqen.com]
Sent: Wednesday, June 22, 2011 12:17 PM
To: FreeSWITCH Users Help; Hector Geraldino
Subject: Re: [Freeswitch-users] FS and ASR engine

You might check out http://Nexiwave.com
They have been active at Cluecon and have a web API
that does fully hosted ASR on WAV's or MP3's.

They are doing innovation around running their ASR in GPU clusters,
which means their internal cost of operation is likely to be the lowest in
the industry.

Good ASR is always going to be computationally intensive,
so it is helpful to have somebody managing that for you.
It's going to be a lot easier that trying to juggle your own sphinx training sets.

http://nexiwave.com/index.php/pricing

  --pehr
On Wed, Jun 22, 2011 at 9:54 AM, Hector Geraldino <Hector.Geraldino at ip-soft.net<mailto:Hector.Geraldino at ip-soft.net>> wrote:
Hi everyone,

I want to check with you guys to see if anyone has experience integrating FreeSwitch with an ASR engine, and using the engine as a merely transcriber of the conversation.

I've been playing for the past two weeks or so with Nuance Speech Server/recognizer and pocketsphinx. Nuance is by far a better solution, but due to the lack of freely available documentation and my short expertise in this subject, I haven't been to achieve my goal.

The communication between FS and the ASR engine works great using MCRP, my concern is with the ASR engine itself. I want to allow the user to speak freely, and get a transcription of what the user said. I don't want or need to understand what the meaning of the utterances are (definitely the engine doesn't need to do that), also I don't need/want to write any complex grammar or SLM to get an interpretation of the spoken phrases, I just want the plain text of what has been said. No decisions will be taken based on what the user said, this information will just be passed to a 3rd application.

I don't know if this can be achieved or not without developing grammars (not suitable for open-ended dialogs) or training statistical language models. What I do recall is using Dragon Speak in MS word for dictation, without the need of doing some trtraining or developing grammars. That's exactly what I'm pursuing: a simple plain text transcription of the spoken words.

Have anyone of you deal with something like this by any chance?

Thanks for your help. I apologize if this is not the right place to ask this type of questions.

Thanks again,
Hector

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

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



--
Pehr Anderson, VP Platform Technology
HarQen - http://HarQen.com
414-755-1962 x114
pehr at harqen.com<mailto:pehr at harqen.com>
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From msc at freeswitch.org  Wed Jun 22 20:51:17 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 22 Jun 2011 09:51:17 -0700
Subject: [Freeswitch-users] Need help detecting when phone call is
 answered in lua script (it is starting before the call is picked up)
In-Reply-To: <4E02008B.7080908@499x.com>
References: <4E02008B.7080908@499x.com>
Message-ID: <BANLkTikf8RoG0a-EfG+C_epMD=Xp0A2XEQ@mail.gmail.com>

Wes,

The "originate" API is very similar to the "bridge" dp command - when the
far end sends *any* media then the originate (or bridge) is considered
"successful" and processing continues. You have a few choices, but I would
start by trying this on your originate:

originate {ignore_early_media=true}sofia/gateway/mcw/914145551212 &lua(
hello.lua)

This will tell the originate not to be "successful" until the far end
actually answers, as opposed to when the far end simply sends media. The
drawback here, of course, is that if call progress is sent in band, e.g. a
busy signal, you are explicitly ignoring that information. An alternative is
found in the "execute_on_xxx" family of channel variables. Check it out:
http://wiki.freeswitch.org/wiki/Channel_Variables#The_execute_on_family

It requires a little more thought, but you end up with more fine-tuned
control over everything. Look in particular at the originate & dialplan
example under "execute_on_answer" which I believe contains the most complete
solution to your problem.

-MC

On Wed, Jun 22, 2011 at 7:47 AM, Wes <wes-fs at 499x.com> wrote:

> Hello,
>
> I'm having trouble detecting when a phone call (originated from
> freeswitch) is answered by the called party using a lua script.  I have
> freeswitch connected to a corporate phone system at my office via SIP.
> When I call my deskphone at work (914145551212), the script waits until
> I answer to start playing the welcome message, and other actions.  But
> when I call my cellphone (912625551212), the script starts before I pick
> up the cellphone, as can be seen in the logs I've pasted below.
>
> Even though the lua script isn't properly detecting when the cell phone
> has been answered, freeswitch *does* seem to know, as shown by the line:
> 2011-06-22 09:21:28.334618 [NOTICE] sofia.c:5594 Channel
> [sofia/external/912625551212] has been answered
> which gets logged when I answer the cell, even though the script is
> already running.
>
> This fact gives me hope that there will be a way to fix this script.
>
> Any suggestions are appreciated!
> thanks.
>
> using:
> FreeSWITCH Version 1.0.head (git-8decee3 2011-06-20 13-21-20 -0500)
>
> Script
> http://pastebin.freeswitch.org/16564
>
> logs:
> http://pastebin.freeswitch.org/16563
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From educs13 at yahoo.com.br  Wed Jun 22 20:52:29 2011
From: educs13 at yahoo.com.br (=?utf-8?B?Sm9zw6kgRWR1YXJkbyBkZSBDLiBTaWx2YQ==?=)
Date: Wed, 22 Jun 2011 09:52:29 -0700 (PDT)
Subject: [Freeswitch-users] FS and ASR engine
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD021F939AE4@NY1-EXMB-01.ip-soft.net>
Message-ID: <608380.9320.qm@web161902.mail.bf1.yahoo.com>

Hi Hector,

the point with Dragon Speak is that it already has a language model buil-in (and an acoustic model as well). It's not just an ASR engine. 

To solve your problem, you must have an ASR engine + statistical language model (LM) + acoustic model (AM). These three parts together can be called ASR system and it can be built directly from entreprises like Nuance and Loquendo. But if you want to use a free ASR system you'll have to combine a free engine + a free statistical LM + a free AM.

Two known free engines are Sphinx and Julius. I think that you can find some free english LMs and english AMs supported by these engines at http://www.voxforge.org/ and http://www.keithv.com/software/.

One problem that you'll face with these free engines is the lack of MRCP support. But it can be easily solved just by writing a simple module in FS (for example, see mod_pocketsphinx.c that comes with the source code of FS).

I hope it helps ...
Eduardo


--- Em qua, 22/6/11, Hector Geraldino <Hector.Geraldino at ip-soft.net> escreveu:

De: Hector Geraldino <Hector.Geraldino at ip-soft.net>
Assunto: [Freeswitch-users] FS and ASR engine
Para: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Data: Quarta-feira, 22 de Junho de 2011, 11:54

Hi everyone, ?I want to check with you guys to see if anyone has experience integrating FreeSwitch with an ASR engine, and using the engine as a merely transcriber of the conversation.  ?I?ve been playing for the past two weeks or so with Nuance Speech Server/recognizer and pocketsphinx. Nuance is by far a better solution, but due to the lack of freely available documentation and my short expertise in this subject, I haven?t been to achieve my goal. ?The communication between FS and the ASR engine works great using MCRP, my concern is with the ASR engine itself. I want to allow the user to speak freely, and get a transcription of what the user said. I don?t want or need to understand what the meaning of the utterances are (definitely the engine doesn?t need to do that), also I don?t need/want to write any complex grammar or SLM to get an interpretation of the spoken phrases, I just want the plain text of what has been said. No decisions will
 be taken based on what the user said, this information will just be passed to a 3rd application. ?I don?t know if this can be achieved or not without developing grammars (not suitable for open-ended dialogs) or training statistical language models. What I do recall is using Dragon Speak in MS word for dictation, without the need of doing some trtraining or developing grammars. That?s exactly what I?m pursuing: a simple plain text transcription of the spoken words. ?Have anyone of you deal with something like this by any chance? ?Thanks for your help. I apologize if this is not the right place to ask this type of questions. ?Thanks again,Hector
-----Anexo incorporado-----

_______________________________________________
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From anthony.minessale at gmail.com  Wed Jun 22 20:56:25 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 22 Jun 2011 11:56:25 -0500
Subject: [Freeswitch-users] mod_rtmp client API
In-Reply-To: <F7AC533261894AD2856214A550404626@e1705>
References: <801B6B74CA73484F96B9BBD9BE125561@e1705>
	<BANLkTikmOfC7ot12-sGcO3aZ7+bH_cyGCQ@mail.gmail.com>
	<F7AC533261894AD2856214A550404626@e1705>
Message-ID: <BANLkTi=FSiq_Ls7tKAsTn9pZDhvrcjibeA@mail.gmail.com>

On Wed, Jun 22, 2011 at 11:33 AM, Madovsky <infos at madovsky.org> wrote:
> Ok Anth,
>
> adobe is actually going into the voip market very quickly,
> they started some voip services already.
> so if FS want a chance to create a community for mod_rtmp
> it needs at least a minimum information to use it, if not, I can't see
> the interest of mod_rtmp
>
> Regards

I don't really feel any urgency to speed up the free work we do
because you and adobe are impatient but thanks for the suggestion.



Anthony Minessale II

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

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

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


From bmoore at statirasystems.com  Wed Jun 22 20:34:58 2011
From: bmoore at statirasystems.com (William Moore)
Date: Wed, 22 Jun 2011 12:34:58 -0400
Subject: [Freeswitch-users] Trunk Issue
Message-ID: <4E0219B2.9070808@statirasystems.com>

I can't get any incoming nor outgoing calls. The trunk is through 
Junction Networks and registers fine. I though I had the dial plan 
correct but there is obviously something I am missing.

I am running Blue.Box GUI with FreeSwitch. It is in CentOS 5.5. I am 
using Verizon FiOS internet and the firewall is properly configured. 
This is a test environment so I would be happy to let someone in to take 
a look. I have SSH setup and web access via phone.statirasystems.com. 
Other wise it is a lot of configuration info to post.

I have checked with Junction Networks and everything is fine on there 
end. My server is not responding to the requests.

I am fairly green with freeswitch and would appreciate any help.

Please note: I have sought assistance via the 2600hz community to no avail.

Thank you,

-- 
William J. Moore
Owner
Statira Systems
611 Caroline St
Fredericksburg, VA 22401
540.693.0579
www.statirasystems.com

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From bmoore at statirasystems.com  Wed Jun 22 21:35:04 2011
From: bmoore at statirasystems.com (William Moore)
Date: Wed, 22 Jun 2011 13:35:04 -0400
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <4E0219B2.9070808@statirasystems.com>
References: <4E0219B2.9070808@statirasystems.com>
Message-ID: <4E0227C8.9080304@statirasystems.com>

Some details

/opt/freeswitch/conf/sip_profiles/bluebox_sipinterfaces.xml

        <?xml version="1.0"?>
        <include>
        <profile name="sipinterface_3">
        <settings>
        <param name="enable-timer" value="false"/>
        <param name="user-agent-string" value="Configured by 2600hz!"/>
        <param name="rtp-timer-name" value="soft"/>
        <param name="codec-prefs" value="$${global_codec_prefs}"/>
        <param name="inbound-codec-negotiation" value="generous"/>
        <param name="inbound-reg-force-matching-username" value="true"/>
        <param name="nonce-ttl" value="86400"/>
        <param name="rfc2833-pt" value="101"/>
        <param name="manage-presence" value="true"/>
        <param name="auth-calls" value="false"/>
        <param name="sip-ip" value="$${local_ip_v4}"/>
        <param name="rtp-ip" value="$${local_ip_v4}"/>
        <param name="sip-port" value="5070"/>
        <param name="ext-sip-ip" value="auto-nat"/>
        <param name="ext-rtp-ip" value="auto-nat"/>
        <param name="context" value="multitenant_routing_context"/>
        <param name="apply-nat-acl" value="nat.auto"/>
        </settings>
        </profile>
        <profile name="sipinterface_1">
        <settings>
        <param name="enable-timer" value="false"/>
        <param name="user-agent-string" value="Configured by 2600hz!"/>
        <param name="rtp-timer-name" value="soft"/>
        <param name="codec-prefs" value="$${global_codec_prefs}"/>
        <param name="inbound-codec-negotiation" value="generous"/>
        <param name="inbound-reg-force-matching-username" value="true"/>
        <param name="nonce-ttl" value="86400"/>
        <param name="rfc2833-pt" value="101"/>
        <param name="manage-presence" value="true"/>
        <param name="auth-calls" value="true"/>
        <param name="sip-ip" value="$${local_ip_v4}"/>
        <param name="rtp-ip" value="$${local_ip_v4}"/>
        <param name="sip-port" value="5060"/>
        <param name="nat-options-ping" value="true"/>
        <param name="all-reg-options-ping" value="true"/>
        <param name="ext-sip-ip" value="auto-nat"/>
        <param name="ext-rtp-ip" value="auto-nat"/>
        <param name="context" value="multitenant_routing_context"/>
        <param name="apply-nat-acl" value="nat.auto"/>
        <param name="apply-inbound-acl" value="net_list_5"/>
        <param name="aggressive-nat-detection" value="true"/>
        <param name="log-auth-failures" value="true"/>
        </settings>
        <gateways/>
        </profile>
        <profile name="sipinterface_2">
        <settings>
        <param name="enable-timer" value="false"/>
        <param name="user-agent-string" value="Configured by 2600hz!"/>
        <param name="rtp-timer-name" value="soft"/>
        <param name="codec-prefs" value="$${global_codec_prefs}"/>
        <param name="inbound-codec-negotiation" value="generous"/>
        <param name="inbound-reg-force-matching-username" value="true"/>
        <param name="nonce-ttl" value="86400"/>
        <param name="rfc2833-pt" value="101"/>
        <param name="manage-presence" value="true"/>
        <param name="auth-calls" value="true"/>
        <param name="sip-ip" value="$${local_ip_v4}"/>
        <param name="rtp-ip" value="$${local_ip_v4}"/>
        <param name="sip-port" value="5080"/>
        <param name="nat-options-ping" value="true"/>
        <param name="all-reg-options-ping" value="true"/>
        <param name="ext-sip-ip" value="auto-nat"/>
        <param name="ext-rtp-ip" value="auto-nat"/>
        <param name="context" value="multitenant_routing_context"/>
        <param name="apply-inbound-acl" value="net_list_5"/>
        <param name="apply-register-acl" value="rfc1918.auto"/>
        <param name="log-auth-failures" value="true"/>
        <param name="aggressive-nat-detection" value="true"/>
        </settings>
        <aliases>
        <alias name="$${location_1}"/>
        </aliases>
        <gateways>
        <gateway name="trunk_1">
        <param name="realm" value="jnctn.net"/>
        <param name="register-proxy" value="sip.jnctn.net"/>
        <param name="outbound-proxy" value="sip.jnctn.net"/>
        <param name="username" value="mowcow"/>
        <param name="password" value="RQcLt9ypz5DqoLz2"/>
        <param name="register" value="true"/>
        </gateway>
        </gateways>
        <domains>
        <domain name="all" alias="true" parse="false"/>
        </domains>
        </profile>
        </include>


/opt/freeswitch/conf/dialplan/bluebox_dialplan.xml

        <?xml version="1.0"?>
        <include>
        <context name="context_1">
        <extension name="conditioning_callerid" continue="true">
        <condition field="${internal_caller_id_number}" expression="^.+$">
        <action application="set"
        data="effective_caller_id_name=${internal_caller_id_name}"/>
        <action application="set"
        data="effective_caller_id_number=${internal_caller_id_number}"/>
        </condition>
        </extension>
        <extension name="postroute_global" continue="true">
        <condition>
        <action application="hash"
        data="insert/${domain_name}-spymap/${caller_id_number}/${uuid}"/>
        <action application="hash"
        data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/>
        <action application="hash"
        data="insert/${domain_name}-last_dial/global/${uuid}"/>
        <action application="set" data="RFC2822_DATE=${strftime(%a, %d
        %b %Y %T %z)}"/>
        </condition>
        </extension>
        <extension name="main_number_1" continue="true">
        <condition field="destination_number" expression="^2001$">
        <action application="set" bluebox="settingEndBridge"
        data="hangup_after_bridge=true"/>
        <action application="set" bluebox="settingFail"
        data="continue_on_fail=true"/>
        <action application="set" bluebox="autoPlay"
        data="vm_auto_play=false"/>
        <action application="set" bluebox="settingTimeout"
        data="call_timeout=20"/>
        <action application="set" bluebox="settingRing"
        data="ringback=${us-ring}"/>
        <action application="set" bluebox="settingRingXfer"
        data="transfer_ringback=${us-ring}"/>
        <action application="export" bluebox="sipCalleeIdName"
        data="sip_callee_id_name=Account Admin"/>
        <action application="export" bluebox="sipCalleeIdNumber"
        data="sip_callee_id_number=2001"/>
        <action application="bridge"
        data="user/account_admin@$${location_1}"/>
        <action application="answer"/>
        <action application="voicemail" data="default voicemail_1 2001"/>
        <action application="hangup"/>
        </condition>
        </extension>
        <extension name="main_number_3" continue="true">
        <condition field="destination_number" expression="^502$">
        <action application="set" bluebox="settingEndBridge"
        data="hangup_after_bridge=true"/>
        <action application="set" bluebox="settingFail"
        data="continue_on_fail=true"/>
        <action application="set" bluebox="autoPlay"
        data="vm_auto_play=false"/>
        <action application="set" bluebox="settingTimeout"
        data="call_timeout=30"/>
        <action application="set" bluebox="settingRing"
        data="ringback=${us-ring}"/>
        <action application="set" bluebox="settingRingXfer"
        data="transfer_ringback=${us-ring}"/>
        <action application="export" bluebox="sipCalleeIdName"
        data="sip_callee_id_name=Monkey"/>
        <action application="export" bluebox="sipCalleeIdNumber"
        data="sip_callee_id_number=502"/>
        <action application="bridge" data="user/502@$${location_1}"/>
        <action application="answer"/>
        <action application="voicemail" data="default voicemail_1 2001"/>
        <action application="hangup"/>
        </condition>
        </extension>
        <extension name="main_trunk_1_pattern_1" continue="true">
        <condition field="destination_number" break="never"
        expression="^1{0,1}([2-9][0-8][0-9][2-9][0-9]{6})$"
        bluebox="pattern_1_part_1">
        <action application="set" bluebox="prepend" data="prepend="/>
        <action application="set" bluebox="cid_name"
        data="effective_caller_id_name=Statira Systems"/>
        <action application="set" bluebox="cid_number"
        data="effective_caller_id_number=7036526678"/>
        </condition>
        <condition field="${outbound_caller_id_number}"
        expression="^.+$" break="never" bluebox="caller_id">
        <action application="set"
        data="effective_caller_id_name=${outbound_caller_id_name}"/>
        <action application="set"
        data="effective_caller_id_number=${outbound_caller_id_number}"/>
        <action application="export" bluebox="caller_id_field"
        data="sip_cid_type=rpid"/>
        </condition>
        <condition field="destination_number" break="never"
        expression="^1{0,1}([2-9][0-8][0-9][2-9][0-9]{6})$"
        bluebox="pattern_1_part_1_out">
        <action application="bridge" bluebox="out_trunk_1"
        data="sofia/gateway/trunk_1/${prepend}$1"/>
        </condition>
        </extension>
        </context>
        <context name="context_2">
        <extension name="conditioning_callerid" continue="true">
        <condition field="${internal_caller_id_number}" expression="^.+$">
        <action application="set"
        data="effective_caller_id_name=${internal_caller_id_name}"/>
        <action application="set"
        data="effective_caller_id_number=${internal_caller_id_number}"/>
        </condition>
        </extension>
        <extension name="postroute_global" continue="true">
        <condition>
        <action application="hash"
        data="insert/${domain_name}-spymap/${caller_id_number}/${uuid}"/>
        <action application="hash"
        data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/>
        <action application="hash"
        data="insert/${domain_name}-last_dial/global/${uuid}"/>
        <action application="set" data="RFC2822_DATE=${strftime(%a, %d
        %b %Y %T %z)}"/>
        </condition>
        </extension>
        <extension name="main_number_3" continue="true">
        <condition field="destination_number" expression="^502$">
        <action application="set" bluebox="settingEndBridge"
        data="hangup_after_bridge=true"/>
        <action application="set" bluebox="settingFail"
        data="continue_on_fail=true"/>
        <action application="set" bluebox="autoPlay"
        data="vm_auto_play=false"/>
        <action application="set" bluebox="settingTimeout"
        data="call_timeout=30"/>
        <action application="set" bluebox="settingRing"
        data="ringback=${us-ring}"/>
        <action application="set" bluebox="settingRingXfer"
        data="transfer_ringback=${us-ring}"/>
        <action application="export" bluebox="sipCalleeIdName"
        data="sip_callee_id_name=Monkey"/>
        <action application="export" bluebox="sipCalleeIdNumber"
        data="sip_callee_id_number=502"/>
        <action application="bridge" data="user/502@$${location_1}"/>
        <action application="answer"/>
        <action application="voicemail" data="default voicemail_1 2001"/>
        <action application="hangup"/>
        </condition>
        </extension>
        <extension name="main_number_4" continue="true">
        <condition field="destination_number" expression="^17036526678$">
        <action application="set" bluebox="settingTimeout"
        data="call_timeout=30"/>
        <action application="set" bluebox="settingRing"
        data="ringback=${us-ring}"/>
        <action application="set" bluebox="settingRingXfer"
        data="transfer_ringback=${us-ring}"/>
        <action application="export" bluebox="sipCalleeIdName"
        data="sip_callee_id_name=Account Admin"/>
        <action application="export" bluebox="sipCalleeIdNumber"
        data="sip_callee_id_number=17036526678"/>
        <action application="bridge"
        data="user/account_admin@$${location_1}"/>
        <action application="hangup"/>
        </condition>
        </extension>
        <extension name="main_trunk_1_pattern_1" continue="true">
        <condition field="destination_number" break="never"
        expression="^1{0,1}([2-9][0-8][0-9][2-9][0-9]{6})$"
        bluebox="pattern_1_part_1">
        <action application="set" bluebox="prepend" data="prepend="/>
        <action application="set" bluebox="cid_name"
        data="effective_caller_id_name=Statira Systems"/>
        <action application="set" bluebox="cid_number"
        data="effective_caller_id_number=7036526678"/>
        </condition>
        <condition field="${outbound_caller_id_number}"
        expression="^.+$" break="never" bluebox="caller_id">
        <action application="set"
        data="effective_caller_id_name=${outbound_caller_id_name}"/>
        <action application="set"
        data="effective_caller_id_number=${outbound_caller_id_number}"/>
        <action application="export" bluebox="caller_id_field"
        data="sip_cid_type=rpid"/>
        </condition>
        <condition field="destination_number" break="never"
        expression="^1{0,1}([2-9][0-8][0-9][2-9][0-9]{6})$"
        bluebox="pattern_1_part_1_out">
        <action application="bridge" bluebox="out_trunk_1"
        data="sofia/gateway/trunk_1/${prepend}$1"/>
        </condition>
        </extension>
        </context>
        </include>

/opt/freeswitch/conf/dialplan/bluebox_routes.xml

        <?xml version="1.0"?>
        <include>
        <context name="multitenant_routing_context">
        <extension name="route_number_4" continue="true">
        <condition field="destination_number" expression="^17036526678$">
        <action application="set" data="vm-operator-extension=17036526678"/>
        <action application="set" data="force_transfer_context=context_2"/>
        <action application="transfer" data="17036526678 XML context_2"/>
        </condition>
        </extension>
        </context>
        </include>

William J. Moore
Owner
Statira Systems
611 Caroline St
Fredericksburg, VA 22401
540.693.0579
www.statirasystems.com


On 6/22/2011 12:34 PM, William Moore wrote:
> I can't get any incoming nor outgoing calls. The trunk is through 
> Junction Networks and registers fine. I though I had the dial plan 
> correct but there is obviously something I am missing.
>
> I am running Blue.Box GUI with FreeSwitch. It is in CentOS 5.5. I am 
> using Verizon FiOS internet and the firewall is properly configured. 
> This is a test environment so I would be happy to let someone in to 
> take a look. I have SSH setup and web access via 
> phone.statirasystems.com. Other wise it is a lot of configuration info 
> to post.
>
> I have checked with Junction Networks and everything is fine on there 
> end. My server is not responding to the requests.
>
> I am fairly green with freeswitch and would appreciate any help.
>
> Please note: I have sought assistance via the 2600hz community to no 
> avail.
>
> Thank you,
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From infos at madovsky.org  Wed Jun 22 22:16:38 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 22 Jun 2011 14:16:38 -0400
Subject: [Freeswitch-users] mod_rtmp client API
References: <801B6B74CA73484F96B9BBD9BE125561@e1705><BANLkTikmOfC7ot12-sGcO3aZ7+bH_cyGCQ@mail.gmail.com><F7AC533261894AD2856214A550404626@e1705>
	<BANLkTi=FSiq_Ls7tKAsTn9pZDhvrcjibeA@mail.gmail.com>
Message-ID: <1F6D6705ED694D4C8BEA7D99CA3FF100@e1705>

----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, June 22, 2011 12:56 PM
Subject: Re: [Freeswitch-users] mod_rtmp client API


> On Wed, Jun 22, 2011 at 11:33 AM, Madovsky <infos at madovsky.org> wrote:
>> Ok Anth,
>>
>> adobe is actually going into the voip market very quickly,
>> they started some voip services already.
>> so if FS want a chance to create a community for mod_rtmp
>> it needs at least a minimum information to use it, if not, I can't see
>> the interest of mod_rtmp
>>
>> Regards
> 
> I don't really feel any urgency to speed up the free work we do
> because you and adobe are impatient but thanks for the suggestion.

personaly I use my own flash phone since one year now.
my developer knowledger doesn't make me any wage since 2 years.
I personaly not (yet) make money with FS......

but thanks for your answer....


From wes-fs at 499x.com  Wed Jun 22 22:17:35 2011
From: wes-fs at 499x.com (Wes)
Date: Wed, 22 Jun 2011 13:17:35 -0500
Subject: [Freeswitch-users] Need help detecting when phone call is
 answered in lua script (it is starting before the call is picked up)
In-Reply-To: <BANLkTikf8RoG0a-EfG+C_epMD=Xp0A2XEQ@mail.gmail.com>
References: <4E02008B.7080908@499x.com>
	<BANLkTikf8RoG0a-EfG+C_epMD=Xp0A2XEQ@mail.gmail.com>
Message-ID: <4E0231BF.5020704@499x.com>

Thank you! that worked just fine.

On 6/22/2011 11:51 AM, Michael Collins wrote:
> Wes,
>
> The "originate" API is very similar to the "bridge" dp command - when 
> the far end sends *any* media then the originate (or bridge) is 
> considered "successful" and processing continues. You have a few 
> choices, but I would start by trying this on your originate:
>
> originate {ignore_early_media=true}sofia/gateway/mcw/914145551212 
> &lua(hello.lua)
>
> This will tell the originate not to be "successful" until the far end 
> actually answers, as opposed to when the far end simply sends media. 
> The drawback here, of course, is that if call progress is sent in 
> band, e.g. a busy signal, you are explicitly ignoring that 
> information. An alternative is found in the "execute_on_xxx" family of 
> channel variables. Check it out:
> http://wiki.freeswitch.org/wiki/Channel_Variables#The_execute_on_family
>
> It requires a little more thought, but you end up with more fine-tuned 
> control over everything. Look in particular at the originate & 
> dialplan example under "execute_on_answer" which I believe contains 
> the most complete solution to your problem.
>
> -MC
>
> On Wed, Jun 22, 2011 at 7:47 AM, Wes <wes-fs at 499x.com 
> <mailto:wes-fs at 499x.com>> wrote:
>
>     Hello,
>
>     I'm having trouble detecting when a phone call (originated from
>     freeswitch) is answered by the called party using a lua script.  I
>     have
>     freeswitch connected to a corporate phone system at my office via SIP.
>     When I call my deskphone at work (914145551212), the script waits
>     until
>     I answer to start playing the welcome message, and other actions.  But
>     when I call my cellphone (912625551212), the script starts before
>     I pick
>     up the cellphone, as can be seen in the logs I've pasted below.
>
>     Even though the lua script isn't properly detecting when the cell
>     phone
>     has been answered, freeswitch *does* seem to know, as shown by the
>     line:
>     2011-06-22 09:21:28.334618 [NOTICE] sofia.c:5594 Channel
>     [sofia/external/912625551212] has been answered
>     which gets logged when I answer the cell, even though the script is
>     already running.
>
>     This fact gives me hope that there will be a way to fix this script.
>
>     Any suggestions are appreciated!
>     thanks.
>
>     using:
>     FreeSWITCH Version 1.0.head (git-8decee3 2011-06-20 13-21-20 -0500)
>
>     Script
>     http://pastebin.freeswitch.org/16564
>
>     logs:
>     http://pastebin.freeswitch.org/16563
>
>
>     _______________________________________________
>     Join us at ClueCon 2011, Aug 9-11, Chicago
>     http://www.cluecon.com 877-7-4ACLUE
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From bmoore at statirasystems.com  Wed Jun 22 22:27:25 2011
From: bmoore at statirasystems.com (William Moore)
Date: Wed, 22 Jun 2011 14:27:25 -0400
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <4E0227C8.9080304@statirasystems.com>
References: <4E0219B2.9070808@statirasystems.com>
	<4E0227C8.9080304@statirasystems.com>
Message-ID: <4E02340D.2050708@statirasystems.com>

And what sophiatrace reports from my soft phone device 502 calling 
number 7032203446. When I call from outside in nothing is shown on the 
trace, so I am not sure what that means.

    =~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2011.06.22 14:22:40
    =~=~=~=~=~=~=~=~=~=~=~=
    send 695 bytes to udp/[192.168.1.11]:42615 at 20:28:07.684442:
       
    ------------------------------------------------------------------------
        OPTIONS sip:502 at 192.168.1.11:42615;ob SIP/2.0
        Via: SIP/2.0/UDP 192.168.1.254;rport;branch=z9hG4bK4Hyp1U0Djc12F
        Max-Forwards: 70
        From: <sip:mod_sofia at 192.168.1.254:5060>;tag=trm8SvQyg57ga
        To: <sip:502 at phone.statirasystems.com>
        Call-ID: fb2b0eb4-17b0-122f-85bc-0002a5ed0817
        CSeq: 14049563 OPTIONS
        Contact: <sip:mod_sofia at 192.168.1.254:5060>
        User-Agent: Configured by 2600hz!
        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
    REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
        Supported: precondition, path, replaces
        Allow-Events: talk, hold, presence, dialog, line-seize,
    call-info, sla, include-session-description, presence.winfo,
    message-summary, refer
        Content-Length: 0

       
    ------------------------------------------------------------------------
    recv 1188 bytes from udp/[192.168.1.11]:42615 at 20:28:07.755356:
       
    ------------------------------------------------------------------------
        SIP/2.0 200 OK
        Via: SIP/2.0/UDP
    192.168.1.254;rport=5060;received=192.168.1.254;branch=z9hG4bK4Hyp1U0Djc12F
        Call-ID: fb2b0eb4-17b0-122f-85bc-0002a5ed0817
        From: <sip:mod_sofia at 192.168.1.254>;tag=trm8SvQyg57ga
        To: <sip:502 at phone.statirasystems.com>;tag=z9hG4bK4Hyp1U0Djc12F
        CSeq: 14049563 OPTIONS
        Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE,
    NOTIFY, REFER, MESSAGE, OPTIONS
        Accept: application/sdp, application/pidf+xml,
    application/xpidf+xml, application/simple-message-summary,
    message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
        Supported: replaces, 100rel, timer, norefersub
        Allow-Events: presence, message-summary, refer
        User-Agent: CSipSimple r944 / inc-8
        Content-Type: application/sdp
        Content-Length:   427

        v=0
        o=- 3517756265 3517756265 IN IP4 192.168.1.11
        s=pjmedia
        c=IN IP4 192.168.1.11
        t=0 0
        m=audio 49493 RTP/AVP 9 104 103 105 102 0 8 101
        a=rtcp:40021 IN IP4 192.168.1.11
        a=rtpmap:9 G722/8000
        a=rtpmap:104 speex/32000
        a=rtpmap:103 speex/16000
        a=rtpmap:105 iLBC/8000
        a=fmtp:105 mode=30
        a=rtpmap:102 speex/8000
        a=rtpmap:0 PCMU/8000
        a=rtpmap:8 PCMA/8000
        a=sendrecv
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-15
       
    ------------------------------------------------------------------------
    2011-06-22 13:28:15.458709 [DEBUG] switch_nat.c:510 mapped public
    port 5060 protocol UDP to localport 5060
    2011-06-22 13:28:15.574709 [DEBUG] switch_nat.c:510 mapped public
    port 5060 protocol TCP to localport 5060
    2011-06-22 13:28:15.691704 [DEBUG] switch_nat.c:510 mapped public
    port 5070 protocol UDP to localport 5070
    2011-06-22 13:28:15.808694 [DEBUG] switch_nat.c:510 mapped public
    port 5070 protocol TCP to localport 5070
    2011-06-22 13:28:15.925687 [DEBUG] switch_nat.c:510 mapped public
    port 5080 protocol UDP to localport 5080
    2011-06-22 13:28:16.040679 [DEBUG] switch_nat.c:510 mapped public
    port 5080 protocol TCP to localport 5080
    recv 1346 bytes from udp/[192.168.1.11]:42615 at 20:28:31.536924:
       
    ------------------------------------------------------------------------
        INVITE sip:7032203446 at phone.statirasystems.com SIP/2.0
        Via: SIP/2.0/UDP
    192.168.1.11:42615;rport;branch=z9hG4bKPjYU1YgUOgEnTrbjl09SB0FLBTUw3q.4kD
        Max-Forwards: 70
        From:
    <sip:502 at phone.statirasystems.com>;tag=llr4Rj-HnTn9EQhAB8mLWxBLOUkqnap.
        To: <sip:7032203446 at phone.statirasystems.com>
        Contact: <sip:502 at 192.168.1.11:42615;ob>;+sip.ice
        Call-ID: hiSXYJajRDw17Q9BBhV8MiVL6FHepRwl
        CSeq: 23484 INVITE
        Route: <sip:phone.statirasystems.com;transport=udp;lr>
        Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE,
    NOTIFY, REFER, MESSAGE, OPTIONS
        Supported: replaces, 100rel, timer, norefersub
        Session-Expires: 1800
        Min-SE: 90
        User-Agent: CSipSimple r944 / inc-8
        Content-Type: application/sdp
        Content-Length:   616

        v=0
        o=- 3517756289 3517756289 IN IP4 192.168.1.11
        s=pjmedia
        c=IN IP4 192.168.1.11
        t=0 0
        a=X-nat:0
        m=audio 33616 RTP/AVP 9 104 103 105 102 0 8 101
        a=rtcp:38327 IN IP4 192.168.1.11
        a=rtpmap:9 G722/8000
        a=rtpmap:104 speex/32000
        a=rtpmap:103 speex/16000
        a=rtpmap:105 iLBC/8000
        a=fmtp:105 mode=30
        a=rtpmap:102 speex/8000
        a=rtpmap:0 PCMU/8000
        a=rtpmap:8 PCMA/8000
        a=sendrecv
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-15
        a=ice-ufrag:7adc4b5a
        a=ice-pwd:15961ee1
        a=candidate:Hc0a8010b 1 UDP 2130706431 192.168.1.11 33616 typ host
        a=candidate:Hc0a8010b 2 UDP 2130706430 192.168.1.11 38327 typ host
       
    ------------------------------------------------------------------------
    send 363 bytes to udp/[192.168.1.11]:42615 at 20:28:31.537870:
       
    ------------------------------------------------------------------------
        SIP/2.0 100 Trying
        Via: SIP/2.0/UDP
    192.168.1.11:42615;rport=42615;branch=z9hG4bKPjYU1YgUOgEnTrbjl09SB0FLBTUw3q.4kD
        From:
    <sip:502 at phone.statirasystems.com>;tag=llr4Rj-HnTn9EQhAB8mLWxBLOUkqnap.
        To: <sip:7032203446 at phone.statirasystems.com>
        Call-ID: hiSXYJajRDw17Q9BBhV8MiVL6FHepRwl
        CSeq: 23484 INVITE
        User-Agent: Configured by 2600hz!
        Content-Length: 0

       
    ------------------------------------------------------------------------
    2011-06-22 13:28:31.539663 [DEBUG] sofia.c:6551 IP 192.168.1.11
    Rejected by acl "net_list_5". Falling back to Digest auth.
    2011-06-22 13:28:31.539663 [WARNING] sofia_reg.c:1246 SIP auth
    challenge (INVITE) on sofia profile 'sipinterface_1' for
    [7032203446 at phone.statirasystems.com] from ip 192.168.1.11
    send 855 bytes to udp/[192.168.1.11]:42615 at 20:28:31.541775:
       
    ------------------------------------------------------------------------
        SIP/2.0 407 Proxy Authentication Required
        Via: SIP/2.0/UDP
    192.168.1.11:42615;rport=42615;branch=z9hG4bKPjYU1YgUOgEnTrbjl09SB0FLBTUw3q.4kD
        From:
    <sip:502 at phone.statirasystems.com>;tag=llr4Rj-HnTn9EQhAB8mLWxBLOUkqnap.
        To: <sip:7032203446 at phone.statirasystems.com>;tag=U1D1UQ81Dey3N
        Call-ID: hiSXYJajRDw17Q9BBhV8MiVL6FHepRwl
        CSeq: 23484 INVITE
        User-Agent: Configured by 2600hz!
        Accept: application/sdp
        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
    REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
        Supported: precondition, path, replaces
        Allow-Events: talk, hold, presence, dialog, line-seize,
    call-info, sla, include-session-description, presence.winfo,
    message-summary, refer
        Proxy-Authenticate: Digest realm="phone.statirasystems.com",
    nonce="430ea594-c447-42e5-a795-1c5ac884b996", algorithm=MD5, qop="auth"
        Content-Length: 0

       
    ------------------------------------------------------------------------
    recv 445 bytes from udp/[192.168.1.11]:42615 at 20:28:31.544036:
       
    ------------------------------------------------------------------------
        ACK sip:7032203446 at phone.statirasystems.com SIP/2.0
        Via: SIP/2.0/UDP
    192.168.1.11:42615;rport;branch=z9hG4bKPjYU1YgUOgEnTrbjl09SB0FLBTUw3q.4kD
        Max-Forwards: 70
        From:
    <sip:502 at phone.statirasystems.com>;tag=llr4Rj-HnTn9EQhAB8mLWxBLOUkqnap.
        To: <sip:7032203446 at phone.statirasystems.com>;tag=U1D1UQ81Dey3N
        Call-ID: hiSXYJajRDw17Q9BBhV8MiVL6FHepRwl
        CSeq: 23484 ACK
        Route: <sip:phone.statirasystems.com;transport=udp;lr>
        Content-Length:  0

       
    ------------------------------------------------------------------------
    recv 1643 bytes from udp/[192.168.1.11]:42615 at 20:28:31.545117:
       
    ------------------------------------------------------------------------
        INVITE sip:7032203446 at phone.statirasystems.com SIP/2.0
        Via: SIP/2.0/UDP
    192.168.1.11:42615;rport;branch=z9hG4bKPjRaqtNZWIkhvyivUL2g2vkoVMTmdtxGlf
        Max-Forwards: 70
        From:
    <sip:502 at phone.statirasystems.com>;tag=llr4Rj-HnTn9EQhAB8mLWxBLOUkqnap.
        To: <sip:7032203446 at phone.statirasystems.com>
        Contact: <sip:502 at 192.168.1.11:42615;ob>;+sip.ice
        Call-ID: hiSXYJajRDw17Q9BBhV8MiVL6FHepRwl
        CSeq: 23485 INVITE
        Route: <sip:phone.statirasystems.com;transport=udp;lr>
        Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE,
    NOTIFY, REFER, MESSAGE, OPTIONS
        Supported: replaces, 100rel, timer, norefersub
        Session-Expires: 1800
        Min-SE: 90
        User-Agent: CSipSimple r944 / inc-8
        Proxy-Authorization: Digest username="502",
    realm="phone.statirasystems.com",
    nonce="430ea594-c447-42e5-a795-1c5ac884b996",
    uri="sip:7032203446 at phone.statirasystems.com",
    response="95d75cc4f1478333731f960e0e686399", algorithm=MD5,
    cnonce="VIYZuITTZROoTJla3ithxFWRKkW15K3H", qop=auth, nc=00000001
        Content-Type: application/sdp
        Content-Length:   616

        v=0
        o=- 3517756289 3517756289 IN IP4 192.168.1.11
        s=pjmedia
        c=IN IP4 192.168.1.11
        t=0 0
        a=X-nat:0
        m=audio 33616 RTP/AVP 9 104 103 105 102 0 8 101
        a=rtcp:38327 IN IP4 192.168.1.11
        a=rtpmap:9 G722/8000
        a=rtpmap:104 speex/32000
        a=rtpmap:103 speex/16000
        a=rtpmap:105 iLBC/8000
        a=fmtp:105 mode=30
        a=rtpmap:102 speex/8000
        a=rtpmap:0 PCMU/8000
        a=rtpmap:8 PCMA/8000
        a=sendrecv
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-15
        a=ice-ufrag:7adc4b5a
        a=ice-pwd:15961ee1
        a=candidate:Hc0a8010b 1 UDP 2130706431 192.168.1.11 33616 typ host
        a=candidate:Hc0a8010b 2 UDP 2130706430 192.168.1.11 38327 typ host
       
    ------------------------------------------------------------------------
    send 363 bytes to udp/[192.168.1.11]:42615 at 20:28:31.545859:
       
    ------------------------------------------------------------------------
        SIP/2.0 100 Trying
        Via: SIP/2.0/UDP
    192.168.1.11:42615;rport=42615;branch=z9hG4bKPjRaqtNZWIkhvyivUL2g2vkoVMTmdtxGlf
        From:
    <sip:502 at phone.statirasystems.com>;tag=llr4Rj-HnTn9EQhAB8mLWxBLOUkqnap.
        To: <sip:7032203446 at phone.statirasystems.com>
        Call-ID: hiSXYJajRDw17Q9BBhV8MiVL6FHepRwl
        CSeq: 23485 INVITE
        User-Agent: Configured by 2600hz!
        Content-Length: 0

       
    ------------------------------------------------------------------------
    2011-06-22 13:28:31.547460 [DEBUG] sofia.c:6551 IP 192.168.1.11
    Rejected by acl "net_list_5". Falling back to Digest auth.
    2011-06-22 13:28:31.547460 [NOTICE] switch_channel.c:816 New Channel
    sofia/sipinterface_1/502 at phone.statirasystems.com
    [ae454abd-dc4a-42c9-90cc-6b0b332c44f4]
    2011-06-22 13:28:31.547460 [DEBUG] sofia.c:4761 Channel
    sofia/sipinterface_1/502 at phone.statirasystems.com entering state
    [received][100]
    2011-06-22 13:28:31.547460 [DEBUG] sofia.c:4772 Remote SDP:
    v=0

    o=- 3517756289 3517756289 IN IP4 192.168.1.11

    s=pjmedia

    c=IN IP4 192.168.1.11

    t=0 0

    a=X-nat:0

    m=audio 33616 RTP/AVP 9 104 103 105 102 0 8 101

    a=rtpmap:9 G722/8000

    a=rtpmap:104 speex/32000

    a=rtpmap:103 speex/16000

    a=rtpmap:105 iLBC/8000

    a=fmtp:105 mode=30

    a=rtpmap:102 speex/8000

    a=rtpmap:0 PCMU/8000

    a=rtpmap:8 PCMA/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

    a=rtcp:38327 IN IP4 192.168.1.11

    a=ice-ufrag:7adc4b5a

    a=ice-pwd:15961ee1

    a=candidate:Hc0a8010b 1 UDP 2130706431 192.168.1.11 33616 typ host

    a=candidate:Hc0a8010b 2 UDP 2130706430 192.168.1.11 38327 typ host


    2011-06-22 13:28:31.547460 [DEBUG] sofia_glue.c:4669 Audio Codec
    Compare [G722:9:8000:20:64000]/[G7221:115:32000:20:48000]
    2011-06-22 13:28:31.547460 [DEBUG] sofia_glue.c:4669 Audio Codec
    Compare [G722:9:8000:20:64000]/[G7221:107:16000:20:32000]
    2011-06-22 13:28:31.547460 [DEBUG] sofia_glue.c:4669 Audio Codec
    Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000]
    2011-06-22 13:28:31.547460 [DEBUG] sofia_glue.c:2792 Set Codec
    sofia/sipinterface_1/502 at phone.statirasystems.com G722/8000 20 ms
    160 samples 64000 bits
    2011-06-22 13:28:31.547460 [DEBUG] sofia_glue.c:4783 Set 2833 dtmf
    send/recv payload to 101
    2011-06-22 13:28:31.547460 [DEBUG] sofia.c:4943
    (sofia/sipinterface_1/502 at phone.statirasystems.com) State Change
    CS_NEW -> CS_INIT
    2011-06-22 13:28:31.547460 [DEBUG] switch_core_session.c:1116 Send
    signal sofia/sipinterface_1/502 at phone.statirasystems.com [BREAK]
    2011-06-22 13:28:31.553607 [DEBUG] switch_core_state_machine.c:325
    (sofia/sipinterface_1/502 at phone.statirasystems.com) Running State
    Change CS_INIT
    2011-06-22 13:28:31.553607 [DEBUG] switch_core_state_machine.c:361
    (sofia/sipinterface_1/502 at phone.statirasystems.com) State INIT
    2011-06-22 13:28:31.553607 [DEBUG] mod_sofia.c:84
    sofia/sipinterface_1/502 at phone.statirasystems.com SOFIA INIT
    2011-06-22 13:28:31.553607 [DEBUG] mod_sofia.c:124
    (sofia/sipinterface_1/502 at phone.statirasystems.com) State Change
    CS_INIT -> CS_ROUTING
    2011-06-22 13:28:31.553607 [DEBUG] switch_core_session.c:1116 Send
    signal sofia/sipinterface_1/502 at phone.statirasystems.com [BREAK]
    2011-06-22 13:28:31.553607 [DEBUG] switch_core_state_machine.c:361
    (sofia/sipinterface_1/502 at phone.statirasystems.com) State INIT going
    to sleep
    2011-06-22 13:28:31.553607 [DEBUG] switch_core_state_machine.c:325
    (sofia/sipinterface_1/502 at phone.statirasystems.com) Running State
    Change CS_ROUTING
    2011-06-22 13:28:31.553607 [DEBUG] switch_channel.c:1672
    (sofia/sipinterface_1/502 at phone.statirasystems.com) Callstate Change
    DOWN -> RINGING
    2011-06-22 13:28:31.555041 [DEBUG] switch_core_state_machine.c:364
    (sofia/sipinterface_1/502 at phone.statirasystems.com) State ROUTING
    2011-06-22 13:28:31.555041 [DEBUG] mod_sofia.c:147
    sofia/sipinterface_1/502 at phone.statirasystems.com SOFIA ROUTING
    2011-06-22 13:28:31.555041 [DEBUG] switch_core_state_machine.c:77
    sofia/sipinterface_1/502 at phone.statirasystems.com Standard ROUTING
    2011-06-22 13:28:31.555041 [INFO] mod_dialplan_xml.c:331 Processing
    502 <502>->7032203446 in context context_1
    Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com parsing
    [context_1->conditioning_callerid] continue=true
    Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com Regex
    (PASS) [conditioning_callerid]
    ${internal_caller_id_number}(7036526678) =~ /^.+$/ break=on-false
    Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com Action
    set(effective_caller_id_name=${internal_caller_id_name})
    Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com Action
    set(effective_caller_id_number=${internal_caller_id_number})
    Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com parsing
    [context_1->postroute_global] continue=true
    Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com Absolute
    Condition [postroute_global]
    Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com Action
    hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
    Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com Action
    hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})

    Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com Action
    hash(insert/${domain_name}-last_dial/global/${uuid})
    Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com Action
    set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
    Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com parsing
    [context_1->main_number_1] continue=true
    Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com Regex
    (FAIL) [main_number_1] destination_number(7032203446) =~ /^2001$/
    break=on-false
    Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com parsing
    [context_1->main_number_3] continue=true
    Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com Regex
    (FAIL) [main_number_3] destination_number(7032203446) =~ /^502$/
    break=on-false
    Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com parsing
    [context_1->main_trunk_1_pattern_1] continue=true
    Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com Regex
    (PASS) [main_trunk_1_pattern_1] destination_number(7032203446) =~
    /^1{0,1}([2-9][0-8][0-9][2-9][0-9]{6})$/ break=never
    Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com Action
    set(prepend=)
    Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com Action
    set(effective_caller_id_name=Statira Systems)
    Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com Action
    set(effective_caller_id_number=7036526678)
    Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com Regex
    (PASS) [main_trunk_1_pattern_1]
    ${outbound_caller_id_number}(7036526678) =~ /^.+$/ break=never
    Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com Action
    set(effective_caller_id_name=${outbound_caller_id_name})
    Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com Action
    set(effective_caller_id_number=${outbound_caller_id_number})
    Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com Action
    export(sip_cid_type=rpid)
    Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com Regex
    (PASS) [main_trunk_1_pattern_1] destination_number(7032203446) =~
    /^1{0,1}([2-9][0-8][0-9][2-9][0-9]{6})$/ break=never
    Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com Action
    bridge(sofia/gateway/trunk_1/${prepend}7032203446)
    2011-06-22 13:28:31.558679 [DEBUG] switch_core_state_machine.c:119
    (sofia/sipinterface_1/502 at phone.statirasystems.com) State Change
    CS_ROUTING -> CS_EXECUTE
    2011-06-22 13:28:31.558679 [DEBUG] switch_core_session.c:1116 Send
    signal sofia/sipinterface_1/502 at phone.statirasystems.com [BREAK]
    2011-06-22 13:28:31.558679 [DEBUG] switch_core_state_machine.c:364
    (sofia/sipinterface_1/502 at phone.statirasystems.com) State ROUTING
    going to sleep
    2011-06-22 13:28:31.558679 [DEBUG] switch_core_state_machine.c:325
    (sofia/sipinterface_1/502 at phone.statirasystems.com) Running State
    Change CS_EXECUTE
    2011-06-22 13:28:31.558679 [DEBUG] switch_core_state_machine.c:371
    (sofia/sipinterface_1/502 at phone.statirasystems.com) State EXECUTE
    2011-06-22 13:28:31.558679 [DEBUG] mod_sofia.c:240
    sofia/sipinterface_1/502 at phone.statirasystems.com SOFIA EXECUTE
    2011-06-22 13:28:31.558679 [DEBUG] switch_core_state_machine.c:157
    sofia/sipinterface_1/502 at phone.statirasystems.com Standard EXECUTE
    EXECUTE sofia/sipinterface_1/502 at phone.statirasystems.com
    set(effective_caller_id_name=Monkey)
    2011-06-22 13:28:31.559681 [DEBUG] mod_dptools.c:1060
    sofia/sipinterface_1/502 at phone.statirasystems.com SET
    [effective_caller_id_name]=[Monkey]
    EXECUTE sofia/sipinterface_1/502 at phone.statirasystems.com
    set(effective_caller_id_number=7036526678)
    2011-06-22 13:28:31.559681 [DEBUG] mod_dptools.c:1060
    sofia/sipinterface_1/502 at phone.statirasystems.com SET
    [effective_caller_id_number]=[7036526678]
    EXECUTE sofia/sipinterface_1/502 at phone.statirasystems.com
    hash(insert/phone.statirasystems.com-spymap/502/ae454abd-dc4a-42c9-90cc-6b0b332c44f4)
    EXECUTE sofia/sipinterface_1/502 at phone.statirasystems.com
    hash(insert/phone.statirasystems.com-last_dial/502/7032203446)
    EXECUTE sofia/sipinterface_1/502 at phone.statirasystems.com
    hash(insert/phone.statirasystems.com-last_dial/global/ae454abd-dc4a-42c9-90cc-6b0b332c44f4)
    EXECUTE sofia/sipinterface_1/502 at phone.statirasystems.com
    set(RFC2822_DATE=Wed, 22 Jun 2011 13:28:31 -0700)
    2011-06-22 13:28:31.562678 [DEBUG] mod_dptools.c:1060
    sofia/sipinterface_1/502 at phone.statirasystems.com SET
    [RFC2822_DATE]=[Wed, 22 Jun 2011 13:28:31 -0700]
    EXECUTE sofia/sipinterface_1/502 at phone.statirasystems.com set(prepend=)
    2011-06-22 13:28:31.563814 [DEBUG] mod_dptools.c:1060
    sofia/sipinterface_1/502 at phone.statirasystems.com SET [prepend]=[UNDEF]
    EXECUTE sofia/sipinterface_1/502 at phone.statirasystems.com
    set(effective_caller_id_name=Statira Systems)
    2011-06-22 13:28:31.564681 [DEBUG] mod_dptools.c:1060
    sofia/sipinterface_1/502 at phone.statirasystems.com SET
    [effective_caller_id_name]=[Statira Systems]
    EXECUTE sofia/sipinterface_1/502 at phone.statirasystems.com
    set(effective_caller_id_number=7036526678)
    2011-06-22 13:28:31.565679 [DEBUG] mod_dptools.c:1060
    sofia/sipinterface_1/502 at phone.statirasystems.com SET
    [effective_caller_id_number]=[7036526678]
    EXECUTE sofia/sipinterface_1/502 at phone.statirasystems.com
    set(effective_caller_id_name=Monkey)
    2011-06-22 13:28:31.565679 [DEBUG] mod_dptools.c:1060
    sofia/sipinterface_1/502 at phone.statirasystems.com SET
    [effective_caller_id_name]=[Monkey]
    EXECUTE sofia/sipinterface_1/502 at phone.statirasystems.com
    set(effective_caller_id_number=7036526678)
    2011-06-22 13:28:31.566804 [DEBUG] mod_dptools.c:1060
    sofia/sipinterface_1/502 at phone.statirasystems.com SET
    [effective_caller_id_number]=[7036526678]
    EXECUTE sofia/sipinterface_1/502 at phone.statirasystems.com
    export(sip_cid_type=rpid)
    2011-06-22 13:28:31.567678 [DEBUG] switch_channel.c:965 EXPORT
    (export_vars) [sip_cid_type]=[rpid]
    EXECUTE sofia/sipinterface_1/502 at phone.statirasystems.com
    bridge(sofia/gateway/trunk_1/7032203446)
    2011-06-22 13:28:31.568678 [DEBUG] switch_channel.c:922
    sofia/sipinterface_1/502 at phone.statirasystems.com
    EXPORTING[export_vars] [sip_cid_type]=[rpid] to event
    2011-06-22 13:28:31.568678 [DEBUG] switch_ivr_originate.c:1873
    Parsing global variables
    2011-06-22 13:28:31.568678 [ERR] mod_sofia.c:4045 Invalid Gateway
    2011-06-22 13:28:31.568678 [NOTICE] mod_sofia.c:4402 Close Channel
    N/A [CS_NEW]
    2011-06-22 13:28:31.568678 [DEBUG] switch_core_state_machine.c:457
    () Running State Change CS_DESTROY
    2011-06-22 13:28:31.569833 [DEBUG] switch_core_state_machine.c:467
    (N/A) State DESTROY
    2011-06-22 13:28:31.569833 [DEBUG] mod_sofia.c:362 N/A SOFIA DESTROY
    2011-06-22 13:28:31.569833 [DEBUG] switch_core_state_machine.c:467
    (N/A) State DESTROY going to sleep
    2011-06-22 13:28:31.569833 [ERR] switch_ivr_originate.c:2447 Cannot
    create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
    2011-06-22 13:28:31.569833 [DEBUG] switch_ivr_originate.c:3299
    Originate Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]
    2011-06-22 13:28:31.569833 [INFO] mod_dptools.c:2647 Originate
    Failed.  Cause: INVALID_NUMBER_FORMAT
    2011-06-22 13:28:31.569833 [DEBUG] switch_channel.c:2567
    (sofia/sipinterface_1/502 at phone.statirasystems.com) Callstate Change
    RINGING -> HANGUP
    2011-06-22 13:28:31.570693 [NOTICE] mod_dptools.c:2761 Hangup
    sofia/sipinterface_1/502 at phone.statirasystems.com [CS_EXECUTE]
    [INVALID_NUMBER_FORMAT]
    2011-06-22 13:28:31.570693 [DEBUG] switch_channel.c:2583 Send signal
    sofia/sipinterface_1/502 at phone.statirasystems.com [KILL]
    2011-06-22 13:28:31.570693 [DEBUG] switch_core_session.c:1116 Send
    signal sofia/sipinterface_1/502 at phone.statirasystems.com [BREAK]
    2011-06-22 13:28:31.571679 [DEBUG] switch_core_session.c:2060
    sofia/sipinterface_1/502 at phone.statirasystems.com skip receive
    message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
    2011-06-22 13:28:31.571679 [DEBUG] switch_core_state_machine.c:371
    (sofia/sipinterface_1/502 at phone.statirasystems.com) State EXECUTE
    going to sleep
    2011-06-22 13:28:31.571679 [DEBUG] switch_core_state_machine.c:325
    (sofia/sipinterface_1/502 at phone.statirasystems.com) Running State
    Change CS_HANGUP
    2011-06-22 13:28:31.572811 [DEBUG] switch_core_state_machine.c:565
    (sofia/sipinterface_1/502 at phone.statirasystems.com) State HANGUP
    2011-06-22 13:28:31.572811 [DEBUG] mod_sofia.c:457 Channel
    sofia/sipinterface_1/502 at phone.statirasystems.com hanging up, cause:
    INVALID_NUMBER_FORMAT
    2011-06-22 13:28:31.573682 [DEBUG] mod_sofia.c:519 Responding to
    INVITE with: 484
    send 861 bytes to udp/[192.168.1.11]:42615 at 20:28:31.574787:
       
    ------------------------------------------------------------------------
        SIP/2.0 484 Address Incomplete
        Via: SIP/2.0/UDP
    192.168.1.11:42615;rport=42615;branch=z9hG4bKPjRaqtNZWIkhvyivUL2g2vkoVMTmdtxGlf
        From:
    <sip:502 at phone.statirasystems.com>;tag=llr4Rj-HnTn9EQhAB8mLWxBLOUkqnap.
        To: <sip:7032203446 at phone.statirasystems.com>;tag=va7SXjS5aQmpH
        Call-ID: hiSXYJajRDw17Q9BBhV8MiVL6FHepRwl
        CSeq: 23485 INVITE
        User-Agent: Configured by 2600hz!
        Accept: application/sdp
        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
    REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
        Supported: precondition, path, replaces
        Allow-Events: talk, hold, presence, dialog, line-seize,
    call-info, sla, include-session-description, presence.winfo,
    message-summary, refer
        Reason: Q.850;cause=28;text="INVALID_NUMBER_FORMAT"
        Content-Length: 0
        Remote-Party-ID: "7032203446"
    <sip:7032203446 at 192.168.1.254>;party=calling;privacy=off;screen=no

       
    ------------------------------------------------------------------------
    2011-06-22 13:28:31.575063 [DEBUG] switch_core_state_machine.c:46
    sofia/sipinterface_1/502 at phone.statirasystems.com Standard HANGUP,
    cause: INVALID_NUMBER_FORMAT
    2011-06-22 13:28:31.575063 [DEBUG] switch_core_state_machine.c:565
    (sofia/sipinterface_1/502 at phone.statirasystems.com) State HANGUP
    going to sleep
    2011-06-22 13:28:31.575063 [DEBUG] switch_core_state_machine.c:356
    (sofia/sipinterface_1/502 at phone.statirasystems.com) State Change
    CS_HANGUP -> CS_REPORTING
    2011-06-22 13:28:31.575063 [DEBUG] switch_core_session.c:1116 Send
    signal sofia/sipinterface_1/502 at phone.statirasystems.com [BREAK]
    2011-06-22 13:28:31.575063 [DEBUG] switch_core_state_machine.c:325
    (sofia/sipinterface_1/502 at phone.statirasystems.com) Running State
    Change CS_REPORTING
    2011-06-22 13:28:31.575845 [DEBUG] switch_core_state_machine.c:625
    (sofia/sipinterface_1/502 at phone.statirasystems.com) State REPORTING
    2011-06-22 13:28:31.575845 [DEBUG] switch_core_state_machine.c:53
    sofia/sipinterface_1/502 at phone.statirasystems.com Standard
    REPORTING, cause: INVALID_NUMBER_FORMAT
    recv 445 bytes from udp/[192.168.1.11]:42615 at 20:28:31.583298:
       
    ------------------------------------------------------------------------
        ACK sip:7032203446 at phone.statirasystems.com SIP/2.0
        Via: SIP/2.0/UDP
    192.168.1.11:42615;rport;branch=z9hG4bKPjRaqtNZWIkhvyivUL2g2vkoVMTmdtxGlf
        Max-Forwards: 70
        From:
    <sip:502 at phone.statirasystems.com>;tag=llr4Rj-HnTn9EQhAB8mLWxBLOUkqnap.
        To: <sip:7032203446 at phone.statirasystems.com>;tag=va7SXjS5aQmpH
        Call-ID: hiSXYJajRDw17Q9BBhV8MiVL6FHepRwl
        CSeq: 23485 ACK
        Route: <sip:phone.statirasystems.com;transport=udp;lr>
        Content-Length:  0

       
    ------------------------------------------------------------------------
    2011-06-22 13:28:31.584480 [DEBUG] switch_core_state_machine.c:625
    (sofia/sipinterface_1/502 at phone.statirasystems.com) State REPORTING
    going to sleep
    2011-06-22 13:28:31.585691 [DEBUG] switch_core_state_machine.c:350
    (sofia/sipinterface_1/502 at phone.statirasystems.com) State Change
    CS_REPORTING -> CS_DESTROY
    2011-06-22 13:28:31.585691 [DEBUG] switch_core_session.c:1116 Send
    signal sofia/sipinterface_1/502 at phone.statirasystems.com [BREAK]
    2011-06-22 13:28:31.585691 [DEBUG] switch_core_session.c:1288
    Session 8 (sofia/sipinterface_1/502 at phone.statirasystems.com)
    Locked, Waiting on external entities
    2011-06-22 13:28:31.585691 [NOTICE] switch_core_session.c:1306
    Session 8 (sofia/sipinterface_1/502 at phone.statirasystems.com) Ended
    2011-06-22 13:28:31.585691 [NOTICE] switch_core_session.c:1308 Close
    Channel sofia/sipinterface_1/502 at phone.statirasystems.com [CS_DESTROY]
    2011-06-22 13:28:31.585691 [DEBUG] switch_core_state_machine.c:454
    (sofia/sipinterface_1/502 at phone.statirasystems.com) Callstate Change
    HANGUP -> DOWN
    2011-06-22 13:28:31.586867 [DEBUG] switch_core_state_machine.c:457
    (sofia/sipinterface_1/502 at phone.statirasystems.com) Running State
    Change CS_DESTROY
    2011-06-22 13:28:31.586867 [DEBUG] switch_core_state_machine.c:467
    (sofia/sipinterface_1/502 at phone.statirasystems.com) State DESTROY
    2011-06-22 13:28:31.586867 [DEBUG] mod_sofia.c:362
    sofia/sipinterface_1/502 at phone.statirasystems.com SOFIA DESTROY
    2011-06-22 13:28:31.586867 [DEBUG] switch_core_state_machine.c:60
    sofia/sipinterface_1/502 at phone.statirasystems.com Standard DESTROY
    2011-06-22 13:28:31.586867 [DEBUG] switch_core_state_machine.c:467
    (sofia/sipinterface_1/502 at phone.statirasystems.com) State DESTROY
    going to sleep
    send 695 bytes to udp/[192.168.1.11]:42615 at 20:28:37.744017:
       
    ------------------------------------------------------------------------
        OPTIONS sip:502 at 192.168.1.11:42615;ob SIP/2.0
        Via: SIP/2.0/UDP 192.168.1.254;rport;branch=z9hG4bK5tQF3pHHFNQNB
        Max-Forwards: 70
        From: <sip:mod_sofia at 192.168.1.254:5060>;tag=XK0jZDa97Za9c
        To: <sip:502 at phone.statirasystems.com>
        Call-ID: 0d15c90c-17b1-122f-85bc-0002a5ed0817
        CSeq: 14049564 OPTIONS
        Contact: <sip:mod_sofia at 192.168.1.254:5060>
        User-Agent: Configured by 2600hz!
        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
    REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
        Supported: precondition, path, replaces
        Allow-Events: talk, hold, presence, dialog, line-seize,
    call-info, sla, include-session-description, presence.winfo,
    message-summary, refer
        Content-Length: 0

       
    ------------------------------------------------------------------------
    recv 1188 bytes from udp/[192.168.1.11]:42615 at 20:28:37.860253:
       
    ------------------------------------------------------------------------
        SIP/2.0 200 OK
        Via: SIP/2.0/UDP
    192.168.1.254;rport=5060;received=192.168.1.254;branch=z9hG4bK5tQF3pHHFNQNB
        Call-ID: 0d15c90c-17b1-122f-85bc-0002a5ed0817
        From: <sip:mod_sofia at 192.168.1.254>;tag=XK0jZDa97Za9c
        To: <sip:502 at phone.statirasystems.com>;tag=z9hG4bK5tQF3pHHFNQNB
        CSeq: 14049564 OPTIONS
        Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE,
    NOTIFY, REFER, MESSAGE, OPTIONS
        Accept: application/sdp, application/pidf+xml,
    application/xpidf+xml, application/simple-message-summary,
    message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
        Supported: replaces, 100rel, timer, norefersub
        Allow-Events: presence, message-summary, refer
        User-Agent: CSipSimple r944 / inc-8
        Content-Type: application/sdp
        Content-Length:   427

        v=0
        o=- 3517756295 3517756295 IN IP4 192.168.1.11
        s=pjmedia
        c=IN IP4 192.168.1.11
        t=0 0
        m=audio 49493 RTP/AVP 9 104 103 105 102 0 8 101
        a=rtcp:40021 IN IP4 192.168.1.11
        a=rtpmap:9 G722/8000
        a=rtpmap:104 speex/32000
        a=rtpmap:103 speex/16000
        a=rtpmap:105 iLBC/8000
        a=fmtp:105 mode=30
        a=rtpmap:102 speex/8000
        a=rtpmap:0 PCMU/8000
        a=rtpmap:8 PCMA/8000
        a=sendrecv
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-15
       
    ------------------------------------------------------------------------

William J. Moore
Owner
Statira Systems
611 Caroline St
Fredericksburg, VA 22401
540.693.0579
www.statirasystems.com


On 6/22/2011 1:35 PM, William Moore wrote:
> Some details
>
> /opt/freeswitch/conf/sip_profiles/bluebox_sipinterfaces.xml
>
>         <?xml version="1.0"?>
>         <include>
>         <profile name="sipinterface_3">
>         <settings>
>         <param name="enable-timer" value="false"/>
>         <param name="user-agent-string" value="Configured by 2600hz!"/>
>         <param name="rtp-timer-name" value="soft"/>
>         <param name="codec-prefs" value="$${global_codec_prefs}"/>
>         <param name="inbound-codec-negotiation" value="generous"/>
>         <param name="inbound-reg-force-matching-username" value="true"/>
>         <param name="nonce-ttl" value="86400"/>
>         <param name="rfc2833-pt" value="101"/>
>         <param name="manage-presence" value="true"/>
>         <param name="auth-calls" value="false"/>
>         <param name="sip-ip" value="$${local_ip_v4}"/>
>         <param name="rtp-ip" value="$${local_ip_v4}"/>
>         <param name="sip-port" value="5070"/>
>         <param name="ext-sip-ip" value="auto-nat"/>
>         <param name="ext-rtp-ip" value="auto-nat"/>
>         <param name="context" value="multitenant_routing_context"/>
>         <param name="apply-nat-acl" value="nat.auto"/>
>         </settings>
>         </profile>
>         <profile name="sipinterface_1">
>         <settings>
>         <param name="enable-timer" value="false"/>
>         <param name="user-agent-string" value="Configured by 2600hz!"/>
>         <param name="rtp-timer-name" value="soft"/>
>         <param name="codec-prefs" value="$${global_codec_prefs}"/>
>         <param name="inbound-codec-negotiation" value="generous"/>
>         <param name="inbound-reg-force-matching-username" value="true"/>
>         <param name="nonce-ttl" value="86400"/>
>         <param name="rfc2833-pt" value="101"/>
>         <param name="manage-presence" value="true"/>
>         <param name="auth-calls" value="true"/>
>         <param name="sip-ip" value="$${local_ip_v4}"/>
>         <param name="rtp-ip" value="$${local_ip_v4}"/>
>         <param name="sip-port" value="5060"/>
>         <param name="nat-options-ping" value="true"/>
>         <param name="all-reg-options-ping" value="true"/>
>         <param name="ext-sip-ip" value="auto-nat"/>
>         <param name="ext-rtp-ip" value="auto-nat"/>
>         <param name="context" value="multitenant_routing_context"/>
>         <param name="apply-nat-acl" value="nat.auto"/>
>         <param name="apply-inbound-acl" value="net_list_5"/>
>         <param name="aggressive-nat-detection" value="true"/>
>         <param name="log-auth-failures" value="true"/>
>         </settings>
>         <gateways/>
>         </profile>
>         <profile name="sipinterface_2">
>         <settings>
>         <param name="enable-timer" value="false"/>
>         <param name="user-agent-string" value="Configured by 2600hz!"/>
>         <param name="rtp-timer-name" value="soft"/>
>         <param name="codec-prefs" value="$${global_codec_prefs}"/>
>         <param name="inbound-codec-negotiation" value="generous"/>
>         <param name="inbound-reg-force-matching-username" value="true"/>
>         <param name="nonce-ttl" value="86400"/>
>         <param name="rfc2833-pt" value="101"/>
>         <param name="manage-presence" value="true"/>
>         <param name="auth-calls" value="true"/>
>         <param name="sip-ip" value="$${local_ip_v4}"/>
>         <param name="rtp-ip" value="$${local_ip_v4}"/>
>         <param name="sip-port" value="5080"/>
>         <param name="nat-options-ping" value="true"/>
>         <param name="all-reg-options-ping" value="true"/>
>         <param name="ext-sip-ip" value="auto-nat"/>
>         <param name="ext-rtp-ip" value="auto-nat"/>
>         <param name="context" value="multitenant_routing_context"/>
>         <param name="apply-inbound-acl" value="net_list_5"/>
>         <param name="apply-register-acl" value="rfc1918.auto"/>
>         <param name="log-auth-failures" value="true"/>
>         <param name="aggressive-nat-detection" value="true"/>
>         </settings>
>         <aliases>
>         <alias name="$${location_1}"/>
>         </aliases>
>         <gateways>
>         <gateway name="trunk_1">
>         <param name="realm" value="jnctn.net"/>
>         <param name="register-proxy" value="sip.jnctn.net"/>
>         <param name="outbound-proxy" value="sip.jnctn.net"/>
>         <param name="username" value="mowcow"/>
>         <param name="password" value="RQcLt9ypz5DqoLz2"/>
>         <param name="register" value="true"/>
>         </gateway>
>         </gateways>
>         <domains>
>         <domain name="all" alias="true" parse="false"/>
>         </domains>
>         </profile>
>         </include>
>
>
> /opt/freeswitch/conf/dialplan/bluebox_dialplan.xml
>
>         <?xml version="1.0"?>
>         <include>
>         <context name="context_1">
>         <extension name="conditioning_callerid" continue="true">
>         <condition field="${internal_caller_id_number}" expression="^.+$">
>         <action application="set"
>         data="effective_caller_id_name=${internal_caller_id_name}"/>
>         <action application="set"
>         data="effective_caller_id_number=${internal_caller_id_number}"/>
>         </condition>
>         </extension>
>         <extension name="postroute_global" continue="true">
>         <condition>
>         <action application="hash"
>         data="insert/${domain_name}-spymap/${caller_id_number}/${uuid}"/>
>         <action application="hash"
>         data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/>
>         <action application="hash"
>         data="insert/${domain_name}-last_dial/global/${uuid}"/>
>         <action application="set" data="RFC2822_DATE=${strftime(%a, %d
>         %b %Y %T %z)}"/>
>         </condition>
>         </extension>
>         <extension name="main_number_1" continue="true">
>         <condition field="destination_number" expression="^2001$">
>         <action application="set" bluebox="settingEndBridge"
>         data="hangup_after_bridge=true"/>
>         <action application="set" bluebox="settingFail"
>         data="continue_on_fail=true"/>
>         <action application="set" bluebox="autoPlay"
>         data="vm_auto_play=false"/>
>         <action application="set" bluebox="settingTimeout"
>         data="call_timeout=20"/>
>         <action application="set" bluebox="settingRing"
>         data="ringback=${us-ring}"/>
>         <action application="set" bluebox="settingRingXfer"
>         data="transfer_ringback=${us-ring}"/>
>         <action application="export" bluebox="sipCalleeIdName"
>         data="sip_callee_id_name=Account Admin"/>
>         <action application="export" bluebox="sipCalleeIdNumber"
>         data="sip_callee_id_number=2001"/>
>         <action application="bridge"
>         data="user/account_admin@$${location_1}"/>
>         <action application="answer"/>
>         <action application="voicemail" data="default voicemail_1 2001"/>
>         <action application="hangup"/>
>         </condition>
>         </extension>
>         <extension name="main_number_3" continue="true">
>         <condition field="destination_number" expression="^502$">
>         <action application="set" bluebox="settingEndBridge"
>         data="hangup_after_bridge=true"/>
>         <action application="set" bluebox="settingFail"
>         data="continue_on_fail=true"/>
>         <action application="set" bluebox="autoPlay"
>         data="vm_auto_play=false"/>
>         <action application="set" bluebox="settingTimeout"
>         data="call_timeout=30"/>
>         <action application="set" bluebox="settingRing"
>         data="ringback=${us-ring}"/>
>         <action application="set" bluebox="settingRingXfer"
>         data="transfer_ringback=${us-ring}"/>
>         <action application="export" bluebox="sipCalleeIdName"
>         data="sip_callee_id_name=Monkey"/>
>         <action application="export" bluebox="sipCalleeIdNumber"
>         data="sip_callee_id_number=502"/>
>         <action application="bridge" data="user/502@$${location_1}"/>
>         <action application="answer"/>
>         <action application="voicemail" data="default voicemail_1 2001"/>
>         <action application="hangup"/>
>         </condition>
>         </extension>
>         <extension name="main_trunk_1_pattern_1" continue="true">
>         <condition field="destination_number" break="never"
>         expression="^1{0,1}([2-9][0-8][0-9][2-9][0-9]{6})$"
>         bluebox="pattern_1_part_1">
>         <action application="set" bluebox="prepend" data="prepend="/>
>         <action application="set" bluebox="cid_name"
>         data="effective_caller_id_name=Statira Systems"/>
>         <action application="set" bluebox="cid_number"
>         data="effective_caller_id_number=7036526678"/>
>         </condition>
>         <condition field="${outbound_caller_id_number}"
>         expression="^.+$" break="never" bluebox="caller_id">
>         <action application="set"
>         data="effective_caller_id_name=${outbound_caller_id_name}"/>
>         <action application="set"
>         data="effective_caller_id_number=${outbound_caller_id_number}"/>
>         <action application="export" bluebox="caller_id_field"
>         data="sip_cid_type=rpid"/>
>         </condition>
>         <condition field="destination_number" break="never"
>         expression="^1{0,1}([2-9][0-8][0-9][2-9][0-9]{6})$"
>         bluebox="pattern_1_part_1_out">
>         <action application="bridge" bluebox="out_trunk_1"
>         data="sofia/gateway/trunk_1/${prepend}$1"/>
>         </condition>
>         </extension>
>         </context>
>         <context name="context_2">
>         <extension name="conditioning_callerid" continue="true">
>         <condition field="${internal_caller_id_number}" expression="^.+$">
>         <action application="set"
>         data="effective_caller_id_name=${internal_caller_id_name}"/>
>         <action application="set"
>         data="effective_caller_id_number=${internal_caller_id_number}"/>
>         </condition>
>         </extension>
>         <extension name="postroute_global" continue="true">
>         <condition>
>         <action application="hash"
>         data="insert/${domain_name}-spymap/${caller_id_number}/${uuid}"/>
>         <action application="hash"
>         data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/>
>         <action application="hash"
>         data="insert/${domain_name}-last_dial/global/${uuid}"/>
>         <action application="set" data="RFC2822_DATE=${strftime(%a, %d
>         %b %Y %T %z)}"/>
>         </condition>
>         </extension>
>         <extension name="main_number_3" continue="true">
>         <condition field="destination_number" expression="^502$">
>         <action application="set" bluebox="settingEndBridge"
>         data="hangup_after_bridge=true"/>
>         <action application="set" bluebox="settingFail"
>         data="continue_on_fail=true"/>
>         <action application="set" bluebox="autoPlay"
>         data="vm_auto_play=false"/>
>         <action application="set" bluebox="settingTimeout"
>         data="call_timeout=30"/>
>         <action application="set" bluebox="settingRing"
>         data="ringback=${us-ring}"/>
>         <action application="set" bluebox="settingRingXfer"
>         data="transfer_ringback=${us-ring}"/>
>         <action application="export" bluebox="sipCalleeIdName"
>         data="sip_callee_id_name=Monkey"/>
>         <action application="export" bluebox="sipCalleeIdNumber"
>         data="sip_callee_id_number=502"/>
>         <action application="bridge" data="user/502@$${location_1}"/>
>         <action application="answer"/>
>         <action application="voicemail" data="default voicemail_1 2001"/>
>         <action application="hangup"/>
>         </condition>
>         </extension>
>         <extension name="main_number_4" continue="true">
>         <condition field="destination_number" expression="^17036526678$">
>         <action application="set" bluebox="settingTimeout"
>         data="call_timeout=30"/>
>         <action application="set" bluebox="settingRing"
>         data="ringback=${us-ring}"/>
>         <action application="set" bluebox="settingRingXfer"
>         data="transfer_ringback=${us-ring}"/>
>         <action application="export" bluebox="sipCalleeIdName"
>         data="sip_callee_id_name=Account Admin"/>
>         <action application="export" bluebox="sipCalleeIdNumber"
>         data="sip_callee_id_number=17036526678"/>
>         <action application="bridge"
>         data="user/account_admin@$${location_1}"/>
>         <action application="hangup"/>
>         </condition>
>         </extension>
>         <extension name="main_trunk_1_pattern_1" continue="true">
>         <condition field="destination_number" break="never"
>         expression="^1{0,1}([2-9][0-8][0-9][2-9][0-9]{6})$"
>         bluebox="pattern_1_part_1">
>         <action application="set" bluebox="prepend" data="prepend="/>
>         <action application="set" bluebox="cid_name"
>         data="effective_caller_id_name=Statira Systems"/>
>         <action application="set" bluebox="cid_number"
>         data="effective_caller_id_number=7036526678"/>
>         </condition>
>         <condition field="${outbound_caller_id_number}"
>         expression="^.+$" break="never" bluebox="caller_id">
>         <action application="set"
>         data="effective_caller_id_name=${outbound_caller_id_name}"/>
>         <action application="set"
>         data="effective_caller_id_number=${outbound_caller_id_number}"/>
>         <action application="export" bluebox="caller_id_field"
>         data="sip_cid_type=rpid"/>
>         </condition>
>         <condition field="destination_number" break="never"
>         expression="^1{0,1}([2-9][0-8][0-9][2-9][0-9]{6})$"
>         bluebox="pattern_1_part_1_out">
>         <action application="bridge" bluebox="out_trunk_1"
>         data="sofia/gateway/trunk_1/${prepend}$1"/>
>         </condition>
>         </extension>
>         </context>
>         </include>
>
> /opt/freeswitch/conf/dialplan/bluebox_routes.xml
>
>         <?xml version="1.0"?>
>         <include>
>         <context name="multitenant_routing_context">
>         <extension name="route_number_4" continue="true">
>         <condition field="destination_number" expression="^17036526678$">
>         <action application="set"
>         data="vm-operator-extension=17036526678"/>
>         <action application="set"
>         data="force_transfer_context=context_2"/>
>         <action application="transfer" data="17036526678 XML context_2"/>
>         </condition>
>         </extension>
>         </context>
>         </include>
>
> William J. Moore
> Owner
> Statira Systems
> 611 Caroline St
> Fredericksburg, VA 22401
> 540.693.0579
> www.statirasystems.com
>
> On 6/22/2011 12:34 PM, William Moore wrote:
>> I can't get any incoming nor outgoing calls. The trunk is through 
>> Junction Networks and registers fine. I though I had the dial plan 
>> correct but there is obviously something I am missing.
>>
>> I am running Blue.Box GUI with FreeSwitch. It is in CentOS 5.5. I am 
>> using Verizon FiOS internet and the firewall is properly configured. 
>> This is a test environment so I would be happy to let someone in to 
>> take a look. I have SSH setup and web access via 
>> phone.statirasystems.com. Other wise it is a lot of configuration 
>> info to post.
>>
>> I have checked with Junction Networks and everything is fine on there 
>> end. My server is not responding to the requests.
>>
>> I am fairly green with freeswitch and would appreciate any help.
>>
>> Please note: I have sought assistance via the 2600hz community to no 
>> avail.
>>
>> Thank you,
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com  877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From a.luppi at seletech.com  Wed Jun 22 23:28:41 2011
From: a.luppi at seletech.com (Alessandro)
Date: Wed, 22 Jun 2011 21:28:41 +0200
Subject: [Freeswitch-users] Modified Follow me
In-Reply-To: <4E020A11.4080706@seletech.com>
References: <4E00A61C.1060503@seletech.com>
	<BANLkTimSVMrTGJg72JJe5252z7J-RMXSLA@mail.gmail.com>
	<4E020A11.4080706@seletech.com>
Message-ID: <4E024269.8060503@seletech.com>

Solved,

Thanks

Alessandro

Il 22/06/2011 17:28, Alessandro ha scritto:
> Hi,
>
> this is the v_huntgroup_localhost_$EXTENSION.lua
> http://pastebin.freeswitch.org/16565
>
> if the second line is deleted one problem is solved, FS doesn't answer 
> with 200 ok, but answer with ringing.
> But I'm not able to solve the other problem...
> /    if the first extension is reachable and the user hangup before 
> answering (hangup in ringing), the follow me  procedure starts to 
> contact the second extension in the list. I want         that FS     
> start to contact the second phone only if the first extension isn't 
> registered to FS. /
>
> I think is related with the line 47.
>
> Some one can help me?
>
> Thanks
>
> Alessandro Luppi
>
>
>
> Il 22/06/2011 10:42, Christian Benke ha scritto:
>> On 21 June 2011 16:09, Alessandro<a.luppi at seletech.com>  wrote:
>>> Any suggestion?
>> While i can't give you a concrete solution, take a look at the files
>> that are created by fusionpbx, mainly
>> /usr/local/freeswitch/scripts/v_huntgroup_localhost_$EXTENSION.lua -
>> if you dig a bit you might be able to find and fix the problem
>> yourself(And eventually file a bugreport with fusionpbx).
>>
>> hth
>> Christian
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com  877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> FIRST MAIL:
> /Hi,
>
> i'd like to implement this particular behaviour. If I call an 
> extension that isn't registered to FS, FS re-route the call to another 
> destination. If also the second destination in unreachable the call 
> end. I first used the follow me (created with fusionpbx) to implement 
> this feature, but I have the following problem: if the first extension 
> is reachable and the user hangup before answering (hangup in ringing), 
> the follow me procedure starts to contact the second extension in the 
> list. I want that FS start to contact the second phone only if the 
> first extension isn't registered to FS.
> The second problem is that with follow-me FS answer with 200 OK at the 
> invite of the caller bypassing the ringing status, than the procedure 
> of follow-me is transparent to the caller.
> I'd like that the caller stay in ringing status until one of the 
> destination extension answer at the call or the call is closed by FS.
> Any suggestion?
>
> Regards
>
> Alessandro Luppi /
>
>
> -- 
> Ing. Alessandro Luppi
> Software development
> Seletech srl
> Via Collodi 8, 20052 Monza (MI) - Italy
> Tel: +39.039.5962000 - Fax: +39.039.9716905
> email:a.luppi at seletech.com  - Web:www.seletech.com    orwww.seletech.eu


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

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From elijah at crankenstein.com  Wed Jun 22 23:48:06 2011
From: elijah at crankenstein.com (elijah)
Date: Wed, 22 Jun 2011 12:48:06 -0700
Subject: [Freeswitch-users] 'make current' fails to complete
In-Reply-To: <BANLkTin1FXE3KTtmUiYgQ-3qWiFOZkvQCg@mail.gmail.com>
References: <BANLkTikB19zJpVQSeXs8ouPN=zwdXy3zmQ@mail.gmail.com>
	<BANLkTin1FXE3KTtmUiYgQ-3qWiFOZkvQCg@mail.gmail.com>
Message-ID: <BANLkTi=41stVDFPJG7OCzBtw2iFw9BLo0A@mail.gmail.com>

I did make changes. 'git reset --hard' worked. Thanks.

On Tue, Jun 21, 2011 at 6:11 PM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:

> did you make any changes to the code tree? If you did, you'd probably
> want to perform a git stash to save all the changes you made then run
> git stash apply to merge your changes after the pull.
>
> Otherwise, it would be wise to run git reset --hard to 'fix' the
> source tree if you are sure you made no changes to the code at all
>
> On Tue, Jun 21, 2011 at 5:44 PM, elijah <elijah at crankenstein.com> wrote:
> > My attempts to 'make current' against the Git repository are failing
> today.
> > Can you recommend a method by which I can update my FreeSwitch
> installation?
> > Here's the the last bit of console logging:
> > ...
> > make[2]: Leaving directory `/usr/src/freeswitch/libs/esl/ruby'
> > make -C java clean
> > make[2]: Entering directory `/usr/src/freeswitch/libs/esl/java'
> > rm -f *.o *.so *~ *.jar
> > make[2]: Leaving directory `/usr/src/freeswitch/libs/esl/java'
> > make -C managed clean
> > make[2]: Entering directory `/usr/src/freeswitch/libs/esl/managed'
> > rm -f *.o *.so *~
> > make[2]: Leaving directory `/usr/src/freeswitch/libs/esl/managed'
> > make[1]: Leaving directory `/usr/src/freeswitch/libs/esl'
> > make update
> > make[1]: Entering directory `/usr/src/freeswitch'
> > Pulling updates...
> > Updating 5923f71..4bb7683
> > src/mod/applications/mod_dptools/mod_dptools.c: needs update
> > fatal: Entry 'src/mod/applications/mod_dptools/mod_dptools.c' not
> uptodate.
> > Cannot merge.
> > make[1]: *** [update] Error 128
> > make[1]: Leaving directory `/usr/src/freeswitch'
> > make: *** [current] Error 2
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From wes-fs at 499x.com  Wed Jun 22 23:53:51 2011
From: wes-fs at 499x.com (Wes)
Date: Wed, 22 Jun 2011 14:53:51 -0500
Subject: [Freeswitch-users] how to use session:playAndGetDigits with the
 "say" command in lua script?
Message-ID: <4E02484F.90907@499x.com>

I'm trying to convert an example from a dialplan:

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits#Examples

<extension name="play_and_get_digits with say">
   <condition field="destination_number" expression="^(6500)$">
     <action application="play_and_get_digits" data="1 1 1 3000 # say:'press one for technical support' silence_stream://250 res \d+" />
   </condition>
</extension>

into a lua statement like this:

  digits = session:playAndGetDigits(1, 1, 1, 3000, "#", "say:'press one for technicial support'
", "say:'invalid entry'", "\\d+")

but I'm having trouble with the quoting.  The log says:

2011-06-22 14:40:52.614654 [ERR] switch_ivr_play_say.c:1144 Invalid Args

I can't find any lua examples that use both playAndGetDigits and "say"

any suggestions?



From elijah at crankenstein.com  Wed Jun 22 23:57:28 2011
From: elijah at crankenstein.com (elijah)
Date: Wed, 22 Jun 2011 12:57:28 -0700
Subject: [Freeswitch-users] bind_meta_app no longer works on a-leg of
	outbound call
Message-ID: <BANLkTimG+UXnh3Eop3BhN8RpkzN_kgGwCw@mail.gmail.com>

My dialplan scheme to bind #2 no longer works on the a-leg of outbound calls
after updating against git yesterday. Nothing is logged when a user presses
#2. However, pressing un-mapped keys, such as #3 produces:
2011-06-22 12:46:04.598820 [WARNING] switch_ivr_async.c:2938 sofia/internal/
11 at live001.voice.telifi.com Ignoring meta digit '3' not mapped

here's the relevant dialplan items:

      <condition field="destination_number" expression="^9(\d{10})$">
        <action application="set" data="ringback=${us-ring}"/>
        <action application="set" data="transfer_ringback=$${hold_music}"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="set" data="bind_meta_key=#"/>
        <action application="bind_meta_app" data="2 a s
execute_extension::dx2 XML callsdirect"/>
        <action application="bridge" data="sofia/gateway/onesource/$1"/>
      </condition>

    <extension name="bind_receiver_2">
      <condition field="destination_number" expression="^dx2$">
        <action application="answer"/>
        <action application="transfer" data="-both dynamic_conference XML
callsdirect"/>
      </condition>
    </extension>

    <extension name="dynamic_conference">
      <condition field="destination_number"
expression="^dynamic_conference"/>
      <condition field="rdnis" expression="^(dx2)$">
        <action application="answer"/>
        <action application="bind_digit_action"
data="moderator,*1,exec:execute_extension,ASK_FOR_NUMBER__${telifi_call_id}
XML callsdirect"/>
        <action application="bind_digit_action"
data="moderator,*2,exec:execute_extension,CANCEL_LAST_CALL__${telifi_call_id}
XML callsdirect"/>
        <action application="digit_action_set_realm" data="moderator"/>
        <action application="conference" data="${telifi_call_id}@simple"/>
        <anti-action application="log" data="INFO is moderator: 'no'"/>
        <anti-action application="answer"/>
        <anti-action application="conference" data="${telifi_call_id}@simple
"/>
      </condition>
    </extension>
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From anthony.minessale at gmail.com  Thu Jun 23 00:47:57 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 22 Jun 2011 15:47:57 -0500
Subject: [Freeswitch-users] bind_meta_app no longer works on a-leg of
 outbound call
In-Reply-To: <BANLkTimG+UXnh3Eop3BhN8RpkzN_kgGwCw@mail.gmail.com>
References: <BANLkTimG+UXnh3Eop3BhN8RpkzN_kgGwCw@mail.gmail.com>
Message-ID: <BANLkTinV6rdccW6kmGgjmkhO+jj7984nBg@mail.gmail.com>

I just loaded that config on my box and it works ok.


On Wed, Jun 22, 2011 at 2:57 PM, elijah <elijah at crankenstein.com> wrote:
> My dialplan scheme to bind #2 no longer works on the a-leg of outbound calls
> after updating against git yesterday. Nothing is logged when a user presses
> #2. However, pressing un-mapped keys, such as #3 produces:
> 2011-06-22 12:46:04.598820 [WARNING] switch_ivr_async.c:2938
> sofia/internal/11 at live001.voice.telifi.com Ignoring meta digit '3' not
> mapped
> here's the?relevant?dialplan items:
> ? ? ? <condition field="destination_number" expression="^9(\d{10})$">
> ? ? ? ? <action application="set" data="ringback=${us-ring}"/>
> ? ? ? ? <action application="set" data="transfer_ringback=$${hold_music}"/>
> ? ? ? ? <action application="set" data="continue_on_fail=true"/>
> ? ? ? ? <action application="set" data="bind_meta_key=#"/>
> ? ? ? ? <action application="bind_meta_app" data="2 a s
> execute_extension::dx2 XML callsdirect"/>
> ? ? ? ? <action application="bridge" data="sofia/gateway/onesource/$1"/>
> ? ? ? </condition>
> ? ? <extension name="bind_receiver_2">
> ? ? ? <condition field="destination_number" expression="^dx2$">
> ? ? ? ? <action application="answer"/>
> ? ? ? ? <action application="transfer" data="-both dynamic_conference XML
> callsdirect"/>
> ? ? ? </condition>
> ? ? </extension>
> ? ? <extension name="dynamic_conference">
> ? ? ? <condition field="destination_number"
> expression="^dynamic_conference"/>
> ? ? ? <condition field="rdnis" expression="^(dx2)$">
> ? ? ? ? <action application="answer"/>
> ? ? ? ? <action application="bind_digit_action"
> data="moderator,*1,exec:execute_extension,ASK_FOR_NUMBER__${telifi_call_id}
> XML callsdirect"/>
> ? ? ? ? <action application="bind_digit_action"
> data="moderator,*2,exec:execute_extension,CANCEL_LAST_CALL__${telifi_call_id}
> XML callsdirect"/>
> ? ? ? ? <action application="digit_action_set_realm" data="moderator"/>
> ? ? ? ? <action application="conference" data="${telifi_call_id}@simple"/>
> ? ? ? ? <anti-action application="log" data="INFO is moderator: 'no'"/>
> ? ? ? ? <anti-action application="answer"/>
> ? ? ? ? <anti-action application="conference"
> data="${telifi_call_id}@simple"/>
> ? ? ? </condition>
> ? ? </extension>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From msc at freeswitch.org  Thu Jun 23 00:51:17 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 22 Jun 2011 13:51:17 -0700
Subject: [Freeswitch-users] how to use session:playAndGetDigits with the
 "say" command in lua script?
In-Reply-To: <4E02484F.90907@499x.com>
References: <4E02484F.90907@499x.com>
Message-ID: <BANLkTikXmE0V6yxfM-HMUQ0FKOrY=K=GOw@mail.gmail.com>

Can you confirm: are you trying to use TTS or the say engine? These are two
completely different subjects. If you want to use TTS then you need the
"speak" app, not the "say" app:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_speak

-MC

On Wed, Jun 22, 2011 at 12:53 PM, Wes <wes-fs at 499x.com> wrote:

> I'm trying to convert an example from a dialplan:
>
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits#Examples
>
> <extension name="play_and_get_digits with say">
>   <condition field="destination_number" expression="^(6500)$">
>     <action application="play_and_get_digits" data="1 1 1 3000 # say:'press
> one for technical support' silence_stream://250 res \d+" />
>   </condition>
> </extension>
>
> into a lua statement like this:
>
>  digits = session:playAndGetDigits(1, 1, 1, 3000, "#", "say:'press one for
> technicial support'
> ", "say:'invalid entry'", "\\d+")
>
> but I'm having trouble with the quoting.  The log says:
>
> 2011-06-22 14:40:52.614654 [ERR] switch_ivr_play_say.c:1144 Invalid Args
>
> I can't find any lua examples that use both playAndGetDigits and "say"
>
> any suggestions?
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From wes-fs at 499x.com  Thu Jun 23 01:13:55 2011
From: wes-fs at 499x.com (Wes)
Date: Wed, 22 Jun 2011 16:13:55 -0500
Subject: [Freeswitch-users] how to use session:playAndGetDigits with the
 "say" command in lua script?
In-Reply-To: <BANLkTikXmE0V6yxfM-HMUQ0FKOrY=K=GOw@mail.gmail.com>
References: <4E02484F.90907@499x.com>
	<BANLkTikXmE0V6yxfM-HMUQ0FKOrY=K=GOw@mail.gmail.com>
Message-ID: <4E025B13.90701@499x.com>

Sorry, I was thrown off by the dialplan example... I'm trying to use 
speak... I changed my example to try the speak command and I still have 
the same issue with the quotes.

Is it possible to send in a "speak" phrase as the phrase to use in the 
PAGD command?  Or do I have to record a wav file and pass the name of 
the wave file in for that parameter?

On 6/22/2011 3:51 PM, Michael Collins wrote:
> Can you confirm: are you trying to use TTS or the say engine? These 
> are two completely different subjects. If you want to use TTS then you 
> need the "speak" app, not the "say" app:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_speak
>
> -MC
>
> On Wed, Jun 22, 2011 at 12:53 PM, Wes <wes-fs at 499x.com 
> <mailto:wes-fs at 499x.com>> wrote:
>
>     I'm trying to convert an example from a dialplan:
>
>     http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits#Examples
>
>     <extension name="play_and_get_digits with say">
>     <condition field="destination_number" expression="^(6500)$">
>     <action application="play_and_get_digits" data="1 1 1 3000 #
>     say:'press one for technical support' silence_stream://250 res \d+" />
>     </condition>
>     </extension>
>
>     into a lua statement like this:
>
>      digits = session:playAndGetDigits(1, 1, 1, 3000, "#", "say:'press
>     one for technicial support'
>     ", "say:'invalid entry'", "\\d+")
>
>     but I'm having trouble with the quoting.  The log says:
>
>     2011-06-22 14:40:52.614654 [ERR] switch_ivr_play_say.c:1144
>     Invalid Args
>
>     I can't find any lua examples that use both playAndGetDigits and "say"
>
>     any suggestions?
>
>
>     _______________________________________________
>     Join us at ClueCon 2011, Aug 9-11, Chicago
>     http://www.cluecon.com 877-7-4ACLUE
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From wes-fs at 499x.com  Thu Jun 23 02:09:30 2011
From: wes-fs at 499x.com (Wes)
Date: Wed, 22 Jun 2011 17:09:30 -0500
Subject: [Freeswitch-users] how to use session:playAndGetDigits with the
 "say" command in lua script?
In-Reply-To: <4E025B13.90701@499x.com>
References: <4E02484F.90907@499x.com>	<BANLkTikXmE0V6yxfM-HMUQ0FKOrY=K=GOw@mail.gmail.com>
	<4E025B13.90701@499x.com>
Message-ID: <4E02681A.2070703@499x.com>

or maybe we just need a new version of this method called 
"speakAndGetDigits" that specifically takes a couple of string 
parameters for the words to speak, instead of taking the location of the 
wav files...

On 6/22/2011 4:13 PM, Wes wrote:
> Sorry, I was thrown off by the dialplan example... I'm trying to use 
> speak... I changed my example to try the speak command and I still 
> have the same issue with the quotes.
>
> Is it possible to send in a "speak" phrase as the phrase to use in the 
> PAGD command?  Or do I have to record a wav file and pass the name of 
> the wave file in for that parameter?
>
> On 6/22/2011 3:51 PM, Michael Collins wrote:
>> Can you confirm: are you trying to use TTS or the say engine? These 
>> are two completely different subjects. If you want to use TTS then 
>> you need the "speak" app, not the "say" app:
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_speak
>>
>> -MC
>>
>> On Wed, Jun 22, 2011 at 12:53 PM, Wes <wes-fs at 499x.com 
>> <mailto:wes-fs at 499x.com>> wrote:
>>
>>     I'm trying to convert an example from a dialplan:
>>
>>     http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits#Examples
>>
>>     <extension name="play_and_get_digits with say">
>>     <condition field="destination_number" expression="^(6500)$">
>>     <action application="play_and_get_digits" data="1 1 1 3000 #
>>     say:'press one for technical support' silence_stream://250 res
>>     \d+" />
>>     </condition>
>>     </extension>
>>
>>     into a lua statement like this:
>>
>>      digits = session:playAndGetDigits(1, 1, 1, 3000, "#",
>>     "say:'press one for technicial support'
>>     ", "say:'invalid entry'", "\\d+")
>>
>>     but I'm having trouble with the quoting.  The log says:
>>
>>     2011-06-22 14:40:52.614654 [ERR] switch_ivr_play_say.c:1144
>>     Invalid Args
>>
>>     I can't find any lua examples that use both playAndGetDigits and
>>     "say"
>>
>>     any suggestions?
>>
>>
>>     _______________________________________________
>>     Join us at ClueCon 2011, Aug 9-11, Chicago
>>     http://www.cluecon.com 877-7-4ACLUE
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com  877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From elijah at crankenstein.com  Thu Jun 23 02:13:58 2011
From: elijah at crankenstein.com (elijah)
Date: Wed, 22 Jun 2011 15:13:58 -0700
Subject: [Freeswitch-users] bind_meta_app no longer works on a-leg of
 outbound call
In-Reply-To: <BANLkTinV6rdccW6kmGgjmkhO+jj7984nBg@mail.gmail.com>
References: <BANLkTimG+UXnh3Eop3BhN8RpkzN_kgGwCw@mail.gmail.com>
	<BANLkTinV6rdccW6kmGgjmkhO+jj7984nBg@mail.gmail.com>
Message-ID: <BANLkTikAp0rBuJm6j2u9eWnRRZT4vJGXYQ@mail.gmail.com>

Hmmm. This went from working to spontaneously not working for me yesterday.
However, this same command works on the b-leg of inbound calls. There's no
action from standard logging. What's the best logging setup to catch these
bind commands?


On Wed, Jun 22, 2011 at 1:47 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> I just loaded that config on my box and it works ok.
>
>
> On Wed, Jun 22, 2011 at 2:57 PM, elijah <elijah at crankenstein.com> wrote:
> > My dialplan scheme to bind #2 no longer works on the a-leg of outbound
> calls
> > after updating against git yesterday. Nothing is logged when a user
> presses
> > #2. However, pressing un-mapped keys, such as #3 produces:
> > 2011-06-22 12:46:04.598820 [WARNING] switch_ivr_async.c:2938
> > sofia/internal/11 at live001.voice.telifi.com Ignoring meta digit '3' not
> > mapped
> > here's the relevant dialplan items:
> >       <condition field="destination_number" expression="^9(\d{10})$">
> >         <action application="set" data="ringback=${us-ring}"/>
> >         <action application="set"
> data="transfer_ringback=$${hold_music}"/>
> >         <action application="set" data="continue_on_fail=true"/>
> >         <action application="set" data="bind_meta_key=#"/>
> >         <action application="bind_meta_app" data="2 a s
> > execute_extension::dx2 XML callsdirect"/>
> >         <action application="bridge" data="sofia/gateway/onesource/$1"/>
> >       </condition>
> >     <extension name="bind_receiver_2">
> >       <condition field="destination_number" expression="^dx2$">
> >         <action application="answer"/>
> >         <action application="transfer" data="-both dynamic_conference XML
> > callsdirect"/>
> >       </condition>
> >     </extension>
> >     <extension name="dynamic_conference">
> >       <condition field="destination_number"
> > expression="^dynamic_conference"/>
> >       <condition field="rdnis" expression="^(dx2)$">
> >         <action application="answer"/>
> >         <action application="bind_digit_action"
> >
> data="moderator,*1,exec:execute_extension,ASK_FOR_NUMBER__${telifi_call_id}
> > XML callsdirect"/>
> >         <action application="bind_digit_action"
> >
> data="moderator,*2,exec:execute_extension,CANCEL_LAST_CALL__${telifi_call_id}
> > XML callsdirect"/>
> >         <action application="digit_action_set_realm" data="moderator"/>
> >         <action application="conference" data="${telifi_call_id}@simple
> "/>
> >         <anti-action application="log" data="INFO is moderator: 'no'"/>
> >         <anti-action application="answer"/>
> >         <anti-action application="conference"
> > data="${telifi_call_id}@simple"/>
> >       </condition>
> >     </extension>
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
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From elijah at crankenstein.com  Thu Jun 23 02:53:46 2011
From: elijah at crankenstein.com (elijah)
Date: Wed, 22 Jun 2011 15:53:46 -0700
Subject: [Freeswitch-users] second member to dynamic conference hears only
	hold music
Message-ID: <BANLkTikxRYXfOELxVNV9gD6Wx=Y9SN4=vA@mail.gmail.com>

This setup was working for me until updating against git yesterday. Via a
bind statement on the b-leg of inbound calls, a user is able to pull both
legs into a conference bridge. The problem now is that the a-leg hears only
hold music. I'm having this and another problem with my dialplan after
updating, w/o changing my own configuration - could there be updates to
source that would be affecting me? Here's the setup:

    <extension name="telifi_sales_queue">
      <condition field="destination_number"
expression="^telifi_sales_queue$">
        <action application="bridge_export"
data="telifi_call_id=${telifi_call_id}"/>
        <action application="set" data="bind_meta_key=#"/>
        <action application="bind_meta_app" data="1 b s
execute_extension::dx1 XML callsdirect"/>
        <action application="bind_meta_app" data="2 b s
execute_extension::dx2 XML callsdirect"/>
        <action application="callcenter" data="${
telifi_queue_name}@live001.voice.telifi.com"/>
      </condition>
    </extension>

    <extension name="bind_receiver_2">
      <condition field="destination_number" expression="^dx2$">
        <action application="answer"/>
        <action application="transfer" data="-both dynamic_conference XML
callsdirect"/>
      </condition>
    </extension>

    <extension name="dynamic_conference">
      <condition field="destination_number"
expression="^dynamic_conference"/>
      <condition field="rdnis" expression="^(dx2)$">
        <action application="answer"/>
        <action application="bind_digit_action"
data="moderator,*1,exec:execute_extension,ASK_FOR_NUMBER__${telifi_call_id}
XML callsdirect"/>
        <action application="bind_digit_action"
data="moderator,*2,exec:execute_extension,CANCEL_LAST_CALL__${telifi_call_id}
XML callsdirect"/>
        <action application="digit_action_set_realm" data="moderator"/>
        <action application="conference" data="${telifi_call_id}@simple"/>
        <anti-action application="answer"/>
        <anti-action application="conference" data="${telifi_call_id}@simple
"/>
      </condition>
    </extension>
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From elijah at crankenstein.com  Thu Jun 23 03:00:33 2011
From: elijah at crankenstein.com (elijah)
Date: Wed, 22 Jun 2011 16:00:33 -0700
Subject: [Freeswitch-users] bind_meta_app no longer works on a-leg of
 outbound call
In-Reply-To: <BANLkTikAp0rBuJm6j2u9eWnRRZT4vJGXYQ@mail.gmail.com>
References: <BANLkTimG+UXnh3Eop3BhN8RpkzN_kgGwCw@mail.gmail.com>
	<BANLkTinV6rdccW6kmGgjmkhO+jj7984nBg@mail.gmail.com>
	<BANLkTikAp0rBuJm6j2u9eWnRRZT4vJGXYQ@mail.gmail.com>
Message-ID: <BANLkTi=Hkoq94DFJNrk-DZxdX5cNkW_Y9Q@mail.gmail.com>

Oddly, this same command does work on the b-leg of outbound calls, like
this:

      <action application="bind_meta_app" data="1 b s execute_extension::dx1
XML callsdirect"/>
      <action application="bind_meta_app" data="2 b s execute_extension::dx2
XML callsdirect"/>


On Wed, Jun 22, 2011 at 3:13 PM, elijah <elijah at crankenstein.com> wrote:

> Hmmm. This went from working to spontaneously not working for me yesterday.
> However, this same command works on the b-leg of inbound calls. There's no
> action from standard logging. What's the best logging setup to catch these
> bind commands?
>
>
> On Wed, Jun 22, 2011 at 1:47 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> I just loaded that config on my box and it works ok.
>>
>>
>> On Wed, Jun 22, 2011 at 2:57 PM, elijah <elijah at crankenstein.com> wrote:
>> > My dialplan scheme to bind #2 no longer works on the a-leg of outbound
>> calls
>> > after updating against git yesterday. Nothing is logged when a user
>> presses
>> > #2. However, pressing un-mapped keys, such as #3 produces:
>> > 2011-06-22 12:46:04.598820 [WARNING] switch_ivr_async.c:2938
>> > sofia/internal/11 at live001.voice.telifi.com Ignoring meta digit '3' not
>> > mapped
>> > here's the relevant dialplan items:
>> >       <condition field="destination_number" expression="^9(\d{10})$">
>> >         <action application="set" data="ringback=${us-ring}"/>
>> >         <action application="set"
>> data="transfer_ringback=$${hold_music}"/>
>> >         <action application="set" data="continue_on_fail=true"/>
>> >         <action application="set" data="bind_meta_key=#"/>
>> >         <action application="bind_meta_app" data="2 a s
>> > execute_extension::dx2 XML callsdirect"/>
>> >         <action application="bridge" data="sofia/gateway/onesource/$1"/>
>> >       </condition>
>> >     <extension name="bind_receiver_2">
>> >       <condition field="destination_number" expression="^dx2$">
>> >         <action application="answer"/>
>> >         <action application="transfer" data="-both dynamic_conference
>> XML
>> > callsdirect"/>
>> >       </condition>
>> >     </extension>
>> >     <extension name="dynamic_conference">
>> >       <condition field="destination_number"
>> > expression="^dynamic_conference"/>
>> >       <condition field="rdnis" expression="^(dx2)$">
>> >         <action application="answer"/>
>> >         <action application="bind_digit_action"
>> >
>> data="moderator,*1,exec:execute_extension,ASK_FOR_NUMBER__${telifi_call_id}
>> > XML callsdirect"/>
>> >         <action application="bind_digit_action"
>> >
>> data="moderator,*2,exec:execute_extension,CANCEL_LAST_CALL__${telifi_call_id}
>> > XML callsdirect"/>
>> >         <action application="digit_action_set_realm" data="moderator"/>
>> >         <action application="conference" data="${telifi_call_id}@simple
>> "/>
>> >         <anti-action application="log" data="INFO is moderator: 'no'"/>
>> >         <anti-action application="answer"/>
>> >         <anti-action application="conference"
>> > data="${telifi_call_id}@simple"/>
>> >       </condition>
>> >     </extension>
>> >
>> > _______________________________________________
>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>> > http://www.cluecon.com 877-7-4ACLUE
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From benkokakao at gmail.com  Thu Jun 23 03:23:08 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Thu, 23 Jun 2011 01:23:08 +0200
Subject: [Freeswitch-users] Modified Follow me
In-Reply-To: <4E024269.8060503@seletech.com>
References: <4E00A61C.1060503@seletech.com>
	<BANLkTimSVMrTGJg72JJe5252z7J-RMXSLA@mail.gmail.com>
	<4E020A11.4080706@seletech.com> <4E024269.8060503@seletech.com>
Message-ID: <BANLkTi=QMN1+MvT-c15FFR8O+r2ERbdBKA@mail.gmail.com>

On 22 June 2011 21:28, Alessandro <a.luppi at seletech.com> wrote:
> Solved,

Could you post the solution to complete this thread please?
Respectively file a bug-report/feature-improvement-proposal with
fusionpbx?

Thanks,
Christian


From msc at freeswitch.org  Thu Jun 23 03:42:02 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 22 Jun 2011 16:42:02 -0700
Subject: [Freeswitch-users] how to use session:playAndGetDigits with the
 "say" command in lua script?
In-Reply-To: <4E02681A.2070703@499x.com>
References: <4E02484F.90907@499x.com>
	<BANLkTikXmE0V6yxfM-HMUQ0FKOrY=K=GOw@mail.gmail.com>
	<4E025B13.90701@499x.com> <4E02681A.2070703@499x.com>
Message-ID: <BANLkTi=RHa1W7qxXbQ4MTxhN22Vds97zyA@mail.gmail.com>

Have you researched the subject of "phrase macros"? That may be your golden
ticket. If you have the FS book then look at the latter half of ch 6.
Otherwise check out these resources:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_phrase
conf/lang/en/vm/sounds.xml
conf/lang/en/vm/tts.xml

The voicemail system uses phrase macros a ton, and effectively, I might add.
In short, phrase macros let you piece together sound prompts, calls to the
say app, calls to the speak app, calls to the sleep app, etc. I think you
would be most interested in the "speak-text" action.

-MC


On Wed, Jun 22, 2011 at 3:09 PM, Wes <wes-fs at 499x.com> wrote:

> **
> or maybe we just need a new version of this method called
> "speakAndGetDigits" that specifically takes a couple of string parameters
> for the words to speak, instead of taking the location of the wav files...
>
>
> On 6/22/2011 4:13 PM, Wes wrote:
>
> Sorry, I was thrown off by the dialplan example... I'm trying to use
> speak... I changed my example to try the speak command and I still have the
> same issue with the quotes.
>
> Is it possible to send in a "speak" phrase as the phrase to use in the PAGD
> command?  Or do I have to record a wav file and pass the name of the wave
> file in for that parameter?
>
> On 6/22/2011 3:51 PM, Michael Collins wrote:
>
> Can you confirm: are you trying to use TTS or the say engine? These are two
> completely different subjects. If you want to use TTS then you need the
> "speak" app, not the "say" app:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_speak
>
>  -MC
>
> On Wed, Jun 22, 2011 at 12:53 PM, Wes <wes-fs at 499x.com> wrote:
>
>> I'm trying to convert an example from a dialplan:
>>
>>
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits#Examples
>>
>> <extension name="play_and_get_digits with say">
>>   <condition field="destination_number" expression="^(6500)$">
>>     <action application="play_and_get_digits" data="1 1 1 3000 #
>> say:'press one for technical support' silence_stream://250 res \d+" />
>>   </condition>
>> </extension>
>>
>> into a lua statement like this:
>>
>>  digits = session:playAndGetDigits(1, 1, 1, 3000, "#", "say:'press one for
>> technicial support'
>> ", "say:'invalid entry'", "\\d+")
>>
>> but I'm having trouble with the quoting.  The log says:
>>
>> 2011-06-22 14:40:52.614654 [ERR] switch_ivr_play_say.c:1144 Invalid Args
>>
>> I can't find any lua examples that use both playAndGetDigits and "say"
>>
>> any suggestions?
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Jun 23 03:45:45 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 22 Jun 2011 16:45:45 -0700
Subject: [Freeswitch-users] bind_meta_app no longer works on a-leg of
 outbound call
In-Reply-To: <BANLkTi=Hkoq94DFJNrk-DZxdX5cNkW_Y9Q@mail.gmail.com>
References: <BANLkTimG+UXnh3Eop3BhN8RpkzN_kgGwCw@mail.gmail.com>
	<BANLkTinV6rdccW6kmGgjmkhO+jj7984nBg@mail.gmail.com>
	<BANLkTikAp0rBuJm6j2u9eWnRRZT4vJGXYQ@mail.gmail.com>
	<BANLkTi=Hkoq94DFJNrk-DZxdX5cNkW_Y9Q@mail.gmail.com>
Message-ID: <BANLkTinLMeFKfSmmrPW3ZLckgSk16anViQ@mail.gmail.com>

Have you looked at a pcap of the call for clues? What about DTMF type?
-MC

On Wed, Jun 22, 2011 at 4:00 PM, elijah <elijah at crankenstein.com> wrote:

> Oddly, this same command does work on the b-leg of outbound calls, like
> this:
>
>       <action application="bind_meta_app" data="1 b s
> execute_extension::dx1 XML callsdirect"/>
>       <action application="bind_meta_app" data="2 b s
> execute_extension::dx2 XML callsdirect"/>
>
>
> On Wed, Jun 22, 2011 at 3:13 PM, elijah <elijah at crankenstein.com> wrote:
>
>> Hmmm. This went from working to spontaneously not working for me
>> yesterday. However, this same command works on the b-leg of inbound calls.
>> There's no action from standard logging. What's the best logging setup to
>> catch these bind commands?
>>
>>
>> On Wed, Jun 22, 2011 at 1:47 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> I just loaded that config on my box and it works ok.
>>>
>>>
>>> On Wed, Jun 22, 2011 at 2:57 PM, elijah <elijah at crankenstein.com> wrote:
>>> > My dialplan scheme to bind #2 no longer works on the a-leg of outbound
>>> calls
>>> > after updating against git yesterday. Nothing is logged when a user
>>> presses
>>> > #2. However, pressing un-mapped keys, such as #3 produces:
>>> > 2011-06-22 12:46:04.598820 [WARNING] switch_ivr_async.c:2938
>>> > sofia/internal/11 at live001.voice.telifi.com Ignoring meta digit '3' not
>>> > mapped
>>> > here's the relevant dialplan items:
>>> >       <condition field="destination_number" expression="^9(\d{10})$">
>>> >         <action application="set" data="ringback=${us-ring}"/>
>>> >         <action application="set"
>>> data="transfer_ringback=$${hold_music}"/>
>>> >         <action application="set" data="continue_on_fail=true"/>
>>> >         <action application="set" data="bind_meta_key=#"/>
>>> >         <action application="bind_meta_app" data="2 a s
>>> > execute_extension::dx2 XML callsdirect"/>
>>> >         <action application="bridge"
>>> data="sofia/gateway/onesource/$1"/>
>>> >       </condition>
>>> >     <extension name="bind_receiver_2">
>>> >       <condition field="destination_number" expression="^dx2$">
>>> >         <action application="answer"/>
>>> >         <action application="transfer" data="-both dynamic_conference
>>> XML
>>> > callsdirect"/>
>>> >       </condition>
>>> >     </extension>
>>> >     <extension name="dynamic_conference">
>>> >       <condition field="destination_number"
>>> > expression="^dynamic_conference"/>
>>> >       <condition field="rdnis" expression="^(dx2)$">
>>> >         <action application="answer"/>
>>> >         <action application="bind_digit_action"
>>> >
>>> data="moderator,*1,exec:execute_extension,ASK_FOR_NUMBER__${telifi_call_id}
>>> > XML callsdirect"/>
>>> >         <action application="bind_digit_action"
>>> >
>>> data="moderator,*2,exec:execute_extension,CANCEL_LAST_CALL__${telifi_call_id}
>>> > XML callsdirect"/>
>>> >         <action application="digit_action_set_realm" data="moderator"/>
>>> >         <action application="conference" data="${telifi_call_id}@simple
>>> "/>
>>> >         <anti-action application="log" data="INFO is moderator: 'no'"/>
>>> >         <anti-action application="answer"/>
>>> >         <anti-action application="conference"
>>> > data="${telifi_call_id}@simple"/>
>>> >       </condition>
>>> >     </extension>
>>> >
>>> > _______________________________________________
>>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>>> > http://www.cluecon.com 877-7-4ACLUE
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Jun 23 03:47:38 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 22 Jun 2011 16:47:38 -0700
Subject: [Freeswitch-users] second member to dynamic conference hears
 only hold music
In-Reply-To: <BANLkTikxRYXfOELxVNV9gD6Wx=Y9SN4=vA@mail.gmail.com>
References: <BANLkTikxRYXfOELxVNV9gD6Wx=Y9SN4=vA@mail.gmail.com>
Message-ID: <BANLkTikL34bq5UbzUkvFKcqvZ3D4=XTmAw@mail.gmail.com>

This is the second feature that "broke" for you when updating to latest git.
You may need to clean things out and start over to make sure that you
haven't introduced cruft into the build system. What was your previous
version of FS? What kinds of phones are you using?

On Wed, Jun 22, 2011 at 3:53 PM, elijah <elijah at crankenstein.com> wrote:

> This setup was working for me until updating against git yesterday. Via a
> bind statement on the b-leg of inbound calls, a user is able to pull both
> legs into a conference bridge. The problem now is that the a-leg hears only
> hold music. I'm having this and another problem with my dialplan after
> updating, w/o changing my own configuration - could there be updates to
> source that would be affecting me? Here's the setup:
>
>     <extension name="telifi_sales_queue">
>       <condition field="destination_number"
> expression="^telifi_sales_queue$">
>         <action application="bridge_export"
> data="telifi_call_id=${telifi_call_id}"/>
>         <action application="set" data="bind_meta_key=#"/>
>         <action application="bind_meta_app" data="1 b s
> execute_extension::dx1 XML callsdirect"/>
>         <action application="bind_meta_app" data="2 b s
> execute_extension::dx2 XML callsdirect"/>
>         <action application="callcenter" data="${
> telifi_queue_name}@live001.voice.telifi.com"/>
>       </condition>
>     </extension>
>
>     <extension name="bind_receiver_2">
>       <condition field="destination_number" expression="^dx2$">
>         <action application="answer"/>
>         <action application="transfer" data="-both dynamic_conference XML
> callsdirect"/>
>       </condition>
>     </extension>
>
>     <extension name="dynamic_conference">
>       <condition field="destination_number"
> expression="^dynamic_conference"/>
>       <condition field="rdnis" expression="^(dx2)$">
>         <action application="answer"/>
>         <action application="bind_digit_action"
> data="moderator,*1,exec:execute_extension,ASK_FOR_NUMBER__${telifi_call_id}
> XML callsdirect"/>
>         <action application="bind_digit_action"
> data="moderator,*2,exec:execute_extension,CANCEL_LAST_CALL__${telifi_call_id}
> XML callsdirect"/>
>         <action application="digit_action_set_realm" data="moderator"/>
>         <action application="conference" data="${telifi_call_id}@simple"/>
>         <anti-action application="answer"/>
>         <anti-action application="conference"
> data="${telifi_call_id}@simple"/>
>       </condition>
>     </extension>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Jun 23 03:52:07 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 22 Jun 2011 16:52:07 -0700
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <4E02340D.2050708@statirasystems.com>
References: <4E0219B2.9070808@statirasystems.com>
	<4E0227C8.9080304@statirasystems.com>
	<4E02340D.2050708@statirasystems.com>
Message-ID: <BANLkTiksPsLhd23O-ws626svwnKcJJRmXw@mail.gmail.com>

On Wed, Jun 22, 2011 at 11:27 AM, William Moore
<bmoore at statirasystems.com>wrote:

> **
> And what sophiatrace reports from my soft phone device 502 calling number
> 7032203446. When I call from outside in nothing is shown on the trace, so I
> am not sure what that means.
>

I'm thinking this might be a problem:
2011-06-22 13:28:31.568678 [ERR] mod_sofia.c:4045 Invalid Gateway

Do a "sofia status" and see what shows up for that gateway. Make sure that
gateway is up and reachable. If you have the "options" param enabled in your
gw config then that gw must respond or mod_sofia will assume the gw is
down.

-MC
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From brad at tech21.com  Thu Jun 23 04:19:25 2011
From: brad at tech21.com (Brad Mina)
Date: Wed, 22 Jun 2011 17:19:25 -0700
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <BANLkTiksPsLhd23O-ws626svwnKcJJRmXw@mail.gmail.com>
References: <4E0219B2.9070808@statirasystems.com>
	<4E0227C8.9080304@statirasystems.com>
	<4E02340D.2050708@statirasystems.com>
	<BANLkTiksPsLhd23O-ws626svwnKcJJRmXw@mail.gmail.com>
Message-ID: <BANLkTinAjxG3Bp7jy3GTRs1km_DdoFDrCA@mail.gmail.com>

Also, I'm seeing this being a problem:
   ------------------------------------------------------------------------
   *SIP/2.0 484 Address Incomplete*
   Via: SIP/2.0/UDP 192.168.1.11:42615
;rport=42615;branch=z9hG4bKPjRaqtNZWIkhvyivUL2g2vkoVMTmdtxGlf
   From: <sip:502 at phone.statirasystems.com><sip:502 at phone.statirasystems.com>
;tag=llr4Rj-HnTn9EQhAB8mLWxBLOUkqnap.
   To: <sip:7032203446 at phone.statirasystems.com><sip:7032203446 at phone.statirasystems.com>
;tag=va7SXjS5aQmpH
   Call-ID: hiSXYJajRDw17Q9BBhV8MiVL6FHepRwl
   CSeq: 23485 INVITE
   User-Agent: Configured by 2600hz!
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: precondition, path, replaces
   Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
  * Reason: Q.850;cause=28;text="INVALID_NUMBER_FORMAT"*
   Content-Length: 0
   Remote-Party-ID: "7032203446"
<sip:7032203446 at 192.168.1.254><sip:7032203446 at 192.168.1.254>
;party=calling;privacy=off;screen=no

On Wed, Jun 22, 2011 at 4:52 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Jun 22, 2011 at 11:27 AM, William Moore <bmoore at statirasystems.com
> > wrote:
>
>> **
>> And what sophiatrace reports from my soft phone device 502 calling number
>> 7032203446. When I call from outside in nothing is shown on the trace, so
>> I am not sure what that means.
>>
>
> I'm thinking this might be a problem:
> 2011-06-22 13:28:31.568678 [ERR] mod_sofia.c:4045 Invalid Gateway
>
> Do a "sofia status" and see what shows up for that gateway. Make sure that
> gateway is up and reachable. If you have the "options" param enabled in your
> gw config then that gw must respond or mod_sofia will assume the gw is
> down.
>
> -MC
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Thu Jun 23 04:30:56 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 22 Jun 2011 19:30:56 -0500
Subject: [Freeswitch-users] second member to dynamic conference hears
 only hold music
In-Reply-To: <BANLkTikxRYXfOELxVNV9gD6Wx=Y9SN4=vA@mail.gmail.com>
References: <BANLkTikxRYXfOELxVNV9gD6Wx=Y9SN4=vA@mail.gmail.com>
Message-ID: <BANLkTi=9TWsFmhnqwpom6CDN8U_v5fQYDA@mail.gmail.com>

could you comment line 1696 of switch_channel.c and
make install_core
then restart and retest?

That is the only one I can think of.

On Wed, Jun 22, 2011 at 5:53 PM, elijah <elijah at crankenstein.com> wrote:
> This setup was working for me until updating against git yesterday. Via a
> bind statement on the b-leg of inbound calls, a user is able to pull both
> legs into a conference bridge. The problem now is that the a-leg hears only
> hold music. I'm having this and another problem with my dialplan after
> updating, w/o changing my own configuration - could there be updates to
> source that would be affecting me? Here's the setup:
> ? ? <extension name="telifi_sales_queue">
> ? ? ? <condition field="destination_number"
> expression="^telifi_sales_queue$">
> ? ? ? ? <action application="bridge_export"
> data="telifi_call_id=${telifi_call_id}"/>
> ? ? ? ? <action application="set" data="bind_meta_key=#"/>
> ? ? ? ? <action application="bind_meta_app" data="1 b s
> execute_extension::dx1 XML callsdirect"/>
> ? ? ? ? <action application="bind_meta_app" data="2 b s
> execute_extension::dx2 XML callsdirect"/>
> ? ? ? ? <action application="callcenter"
> data="${telifi_queue_name}@live001.voice.telifi.com"/>
> ? ? ? </condition>
> ? ? </extension>
> ? ? <extension name="bind_receiver_2">
> ? ? ? <condition field="destination_number" expression="^dx2$">
> ? ? ? ? <action application="answer"/>
> ? ? ? ? <action application="transfer" data="-both dynamic_conference XML
> callsdirect"/>
> ? ? ? </condition>
> ? ? </extension>
> ? ? <extension name="dynamic_conference">
> ? ? ? <condition field="destination_number"
> expression="^dynamic_conference"/>
> ? ? ? <condition field="rdnis" expression="^(dx2)$">
> ? ? ? ? <action application="answer"/>
> ? ? ? ? <action application="bind_digit_action"
> data="moderator,*1,exec:execute_extension,ASK_FOR_NUMBER__${telifi_call_id}
> XML callsdirect"/>
> ? ? ? ? <action application="bind_digit_action"
> data="moderator,*2,exec:execute_extension,CANCEL_LAST_CALL__${telifi_call_id}
> XML callsdirect"/>
> ? ? ? ? <action application="digit_action_set_realm" data="moderator"/>
> ? ? ? ? <action application="conference" data="${telifi_call_id}@simple"/>
> ? ? ? ? <anti-action application="answer"/>
> ? ? ? ? <anti-action application="conference"
> data="${telifi_call_id}@simple"/>
> ? ? ? </condition>
> ? ? </extension>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From msc at freeswitch.org  Thu Jun 23 04:31:16 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 22 Jun 2011 17:31:16 -0700
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <BANLkTinAjxG3Bp7jy3GTRs1km_DdoFDrCA@mail.gmail.com>
References: <4E0219B2.9070808@statirasystems.com>
	<4E0227C8.9080304@statirasystems.com>
	<4E02340D.2050708@statirasystems.com>
	<BANLkTiksPsLhd23O-ws626svwnKcJJRmXw@mail.gmail.com>
	<BANLkTinAjxG3Bp7jy3GTRs1km_DdoFDrCA@mail.gmail.com>
Message-ID: <BANLkTimip1BOfOG+U9rBOAJeiUJK_PL4fw@mail.gmail.com>

The 484 I believe is a result of the bad gateway. In any case, the OP needs
to figure out what's up w/ that gw or no calls will be going out...

-MC

On Wed, Jun 22, 2011 at 5:19 PM, Brad Mina <brad at tech21.com> wrote:

> Also, I'm seeing this being a problem:
>    ------------------------------------------------------------------------
>    *SIP/2.0 484 Address Incomplete*
>    Via: SIP/2.0/UDP 192.168.1.11:42615
> ;rport=42615;branch=z9hG4bKPjRaqtNZWIkhvyivUL2g2vkoVMTmdtxGlf
>    From: <sip:502 at phone.statirasystems.com><sip:502 at phone.statirasystems.com>
> ;tag=llr4Rj-HnTn9EQhAB8mLWxBLOUkqnap.
>    To: <sip:7032203446 at phone.statirasystems.com><sip:7032203446 at phone.statirasystems.com>
> ;tag=va7SXjS5aQmpH
>    Call-ID: hiSXYJajRDw17Q9BBhV8MiVL6FHepRwl
>    CSeq: 23485 INVITE
>    User-Agent: Configured by 2600hz!
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: precondition, path, replaces
>    Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>   * Reason: Q.850;cause=28;text="INVALID_NUMBER_FORMAT"*
>    Content-Length: 0
>    Remote-Party-ID: "7032203446" <sip:7032203446 at 192.168.1.254><sip:7032203446 at 192.168.1.254>
> ;party=calling;privacy=off;screen=no
>
> On Wed, Jun 22, 2011 at 4:52 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Wed, Jun 22, 2011 at 11:27 AM, William Moore <
>> bmoore at statirasystems.com> wrote:
>>
>>> **
>>> And what sophiatrace reports from my soft phone device 502 calling number
>>> 7032203446. When I call from outside in nothing is shown on the trace,
>>> so I am not sure what that means.
>>>
>>
>> I'm thinking this might be a problem:
>> 2011-06-22 13:28:31.568678 [ERR] mod_sofia.c:4045 Invalid Gateway
>>
>> Do a "sofia status" and see what shows up for that gateway. Make sure that
>> gateway is up and reachable. If you have the "options" param enabled in your
>> gw config then that gw must respond or mod_sofia will assume the gw is
>> down.
>>
>> -MC
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brad at tech21.com  Thu Jun 23 04:34:46 2011
From: brad at tech21.com (Brad Mina)
Date: Wed, 22 Jun 2011 17:34:46 -0700
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <BANLkTimip1BOfOG+U9rBOAJeiUJK_PL4fw@mail.gmail.com>
References: <4E0219B2.9070808@statirasystems.com>
	<4E0227C8.9080304@statirasystems.com>
	<4E02340D.2050708@statirasystems.com>
	<BANLkTiksPsLhd23O-ws626svwnKcJJRmXw@mail.gmail.com>
	<BANLkTinAjxG3Bp7jy3GTRs1km_DdoFDrCA@mail.gmail.com>
	<BANLkTimip1BOfOG+U9rBOAJeiUJK_PL4fw@mail.gmail.com>
Message-ID: <BANLkTimBC4FKan=kjNoizbGUFGAe1ACu-A@mail.gmail.com>

Ah, thought it was recv not send. Good to know that an invalid gateway issue
can cause this.

On Wed, Jun 22, 2011 at 5:31 PM, Michael Collins <msc at freeswitch.org> wrote:

> The 484 I believe is a result of the bad gateway. In any case, the OP needs
> to figure out what's up w/ that gw or no calls will be going out...
>
> -MC
>
>
> On Wed, Jun 22, 2011 at 5:19 PM, Brad Mina <brad at tech21.com> wrote:
>
>> Also, I'm seeing this being a problem:
>>
>>  ------------------------------------------------------------------------
>>    *SIP/2.0 484 Address Incomplete*
>>    Via: SIP/2.0/UDP 192.168.1.11:42615
>> ;rport=42615;branch=z9hG4bKPjRaqtNZWIkhvyivUL2g2vkoVMTmdtxGlf
>>    From: <sip:502 at phone.statirasystems.com><sip:502 at phone.statirasystems.com>
>> ;tag=llr4Rj-HnTn9EQhAB8mLWxBLOUkqnap.
>>    To: <sip:7032203446 at phone.statirasystems.com><sip:7032203446 at phone.statirasystems.com>
>> ;tag=va7SXjS5aQmpH
>>    Call-ID: hiSXYJajRDw17Q9BBhV8MiVL6FHepRwl
>>    CSeq: 23485 INVITE
>>    User-Agent: Configured by 2600hz!
>>    Accept: application/sdp
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>    Supported: precondition, path, replaces
>>    Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
>> include-session-description, presence.winfo, message-summary, refer
>>   * Reason: Q.850;cause=28;text="INVALID_NUMBER_FORMAT"*
>>    Content-Length: 0
>>    Remote-Party-ID: "7032203446" <sip:7032203446 at 192.168.1.254><sip:7032203446 at 192.168.1.254>
>> ;party=calling;privacy=off;screen=no
>>
>> On Wed, Jun 22, 2011 at 4:52 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>>
>>>
>>> On Wed, Jun 22, 2011 at 11:27 AM, William Moore <
>>> bmoore at statirasystems.com> wrote:
>>>
>>>> **
>>>> And what sophiatrace reports from my soft phone device 502 calling
>>>> number 7032203446. When I call from outside in nothing is shown on the
>>>> trace, so I am not sure what that means.
>>>>
>>>
>>> I'm thinking this might be a problem:
>>> 2011-06-22 13:28:31.568678 [ERR] mod_sofia.c:4045 Invalid Gateway
>>>
>>> Do a "sofia status" and see what shows up for that gateway. Make sure
>>> that gateway is up and reachable. If you have the "options" param enabled in
>>> your gw config then that gw must respond or mod_sofia will assume the gw is
>>> down.
>>>
>>> -MC
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Thu Jun 23 04:37:59 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 22 Jun 2011 19:37:59 -0500
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <BANLkTimBC4FKan=kjNoizbGUFGAe1ACu-A@mail.gmail.com>
References: <4E0219B2.9070808@statirasystems.com>
	<4E0227C8.9080304@statirasystems.com>
	<4E02340D.2050708@statirasystems.com>
	<BANLkTiksPsLhd23O-ws626svwnKcJJRmXw@mail.gmail.com>
	<BANLkTinAjxG3Bp7jy3GTRs1km_DdoFDrCA@mail.gmail.com>
	<BANLkTimip1BOfOG+U9rBOAJeiUJK_PL4fw@mail.gmail.com>
	<BANLkTimBC4FKan=kjNoizbGUFGAe1ACu-A@mail.gmail.com>
Message-ID: <AC8EC5A4-1799-433A-9DD0-A715ADF5F256@freeswitch.org>

484 is address incomplete means you didn't send enough digits to the gateway usually.

/b

On Jun 22, 2011, at 7:34 PM, Brad Mina wrote:

> Ah, thought it was recv not send. Good to know that an invalid gateway issue
> can cause this.
> 
> On Wed, Jun 22, 2011 at 5:31 PM, Michael Collins <msc at freeswitch.org> wrote:
> 
>> The 484 I believe is a result of the bad gateway. In any case, the OP needs
>> to figure out what's up w/ that gw or no calls will be going out...
>> 
>> -MC
> 

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From elijah at crankenstein.com  Thu Jun 23 05:00:48 2011
From: elijah at crankenstein.com (elijah)
Date: Wed, 22 Jun 2011 18:00:48 -0700
Subject: [Freeswitch-users] second member to dynamic conference hears
 only hold music
In-Reply-To: <BANLkTi=9TWsFmhnqwpom6CDN8U_v5fQYDA@mail.gmail.com>
References: <BANLkTikxRYXfOELxVNV9gD6Wx=Y9SN4=vA@mail.gmail.com>
	<BANLkTi=9TWsFmhnqwpom6CDN8U_v5fQYDA@mail.gmail.com>
Message-ID: <BANLkTi=+nT1LT_8pB_7yy_rMf37TFixwcA@mail.gmail.com>

That didn't help. Maybe I've screwed this up by running 'git reset --hard'?
I've confirmed this version works for me: FreeSWITCH Version 1.0.head
(git-5923f71 2011-06-01 22-36-19 -0500)
and this does not: FreeSWITCH Version 1.0.head (git-288455c 2011-06-22
17-05-53 -0400)

On Wed, Jun 22, 2011 at 5:30 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> could you comment line 1696 of switch_channel.c and
> make install_core
> then restart and retest?
>
> That is the only one I can think of.
>
> On Wed, Jun 22, 2011 at 5:53 PM, elijah <elijah at crankenstein.com> wrote:
> > This setup was working for me until updating against git yesterday. Via a
> > bind statement on the b-leg of inbound calls, a user is able to pull both
> > legs into a conference bridge. The problem now is that the a-leg hears
> only
> > hold music. I'm having this and another problem with my dialplan after
> > updating, w/o changing my own configuration - could there be updates to
> > source that would be affecting me? Here's the setup:
> >     <extension name="telifi_sales_queue">
> >       <condition field="destination_number"
> > expression="^telifi_sales_queue$">
> >         <action application="bridge_export"
> > data="telifi_call_id=${telifi_call_id}"/>
> >         <action application="set" data="bind_meta_key=#"/>
> >         <action application="bind_meta_app" data="1 b s
> > execute_extension::dx1 XML callsdirect"/>
> >         <action application="bind_meta_app" data="2 b s
> > execute_extension::dx2 XML callsdirect"/>
> >         <action application="callcenter"
> > data="${telifi_queue_name}@live001.voice.telifi.com"/>
> >       </condition>
> >     </extension>
> >     <extension name="bind_receiver_2">
> >       <condition field="destination_number" expression="^dx2$">
> >         <action application="answer"/>
> >         <action application="transfer" data="-both dynamic_conference XML
> > callsdirect"/>
> >       </condition>
> >     </extension>
> >     <extension name="dynamic_conference">
> >       <condition field="destination_number"
> > expression="^dynamic_conference"/>
> >       <condition field="rdnis" expression="^(dx2)$">
> >         <action application="answer"/>
> >         <action application="bind_digit_action"
> >
> data="moderator,*1,exec:execute_extension,ASK_FOR_NUMBER__${telifi_call_id}
> > XML callsdirect"/>
> >         <action application="bind_digit_action"
> >
> data="moderator,*2,exec:execute_extension,CANCEL_LAST_CALL__${telifi_call_id}
> > XML callsdirect"/>
> >         <action application="digit_action_set_realm" data="moderator"/>
> >         <action application="conference" data="${telifi_call_id}@simple
> "/>
> >         <anti-action application="answer"/>
> >         <anti-action application="conference"
> > data="${telifi_call_id}@simple"/>
> >       </condition>
> >     </extension>
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From curriegrad2004 at gmail.com  Thu Jun 23 06:02:47 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Wed, 22 Jun 2011 19:02:47 -0700
Subject: [Freeswitch-users] 'make current' fails to complete
In-Reply-To: <BANLkTi=41stVDFPJG7OCzBtw2iFw9BLo0A@mail.gmail.com>
References: <BANLkTikB19zJpVQSeXs8ouPN=zwdXy3zmQ@mail.gmail.com>
	<BANLkTin1FXE3KTtmUiYgQ-3qWiFOZkvQCg@mail.gmail.com>
	<BANLkTi=41stVDFPJG7OCzBtw2iFw9BLo0A@mail.gmail.com>
Message-ID: <BANLkTi=NeD_OJ450AG+cVYV2RY4Lnnsz6w@mail.gmail.com>

Yeah, the next time you make changes to the git tree, you would want
to first stash all your changes by running git stash and then do the
git pull. After that, run git stash apply to restore the changes you
made to the code on the git tree.

On Wed, Jun 22, 2011 at 12:48 PM, elijah <elijah at crankenstein.com> wrote:
> I did make changes. 'git reset --hard' worked. Thanks.
>
> On Tue, Jun 21, 2011 at 6:11 PM, curriegrad2004 <curriegrad2004 at gmail.com>
> wrote:
>>
>> did you make any changes to the code tree? If you did, you'd probably
>> want to perform a git stash to save all the changes you made then run
>> git stash apply to merge your changes after the pull.
>>
>> Otherwise, it would be wise to run git reset --hard to 'fix' the
>> source tree if you are sure you made no changes to the code at all
>>
>> On Tue, Jun 21, 2011 at 5:44 PM, elijah <elijah at crankenstein.com> wrote:
>> > My attempts to 'make current' against the Git repository are failing
>> > today.
>> > Can you?recommend?a method by which I can update my FreeSwitch
>> > installation?
>> > Here's the the last bit of console logging:
>> > ...
>> > make[2]: Leaving directory `/usr/src/freeswitch/libs/esl/ruby'
>> > make -C java clean
>> > make[2]: Entering directory `/usr/src/freeswitch/libs/esl/java'
>> > rm -f *.o *.so *~ *.jar
>> > make[2]: Leaving directory `/usr/src/freeswitch/libs/esl/java'
>> > make -C managed clean
>> > make[2]: Entering directory `/usr/src/freeswitch/libs/esl/managed'
>> > rm -f *.o *.so *~
>> > make[2]: Leaving directory `/usr/src/freeswitch/libs/esl/managed'
>> > make[1]: Leaving directory `/usr/src/freeswitch/libs/esl'
>> > make update
>> > make[1]: Entering directory `/usr/src/freeswitch'
>> > Pulling updates...
>> > Updating 5923f71..4bb7683
>> > src/mod/applications/mod_dptools/mod_dptools.c: needs update
>> > fatal: Entry 'src/mod/applications/mod_dptools/mod_dptools.c' not
>> > uptodate.
>> > Cannot merge.
>> > make[1]: *** [update] Error 128
>> > make[1]: Leaving directory `/usr/src/freeswitch'
>> > make: *** [current] Error 2
>> >
>> > _______________________________________________
>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>> > http://www.cluecon.com 877-7-4ACLUE
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From spencer at 5ninesolutions.com  Thu Jun 23 08:53:11 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Wed, 22 Jun 2011 21:53:11 -0700
Subject: [Freeswitch-users] Linksys SPA509 MWI Subscribe
Message-ID: <EC0F9BE7-14B9-428B-B4AF-2973DE18C1D7@5ninesolutions.com>

Hello all,
I'm trying to set up a shared VM box with a few Cisco/Linksys SPA509G phones.  Does anyone know if it is possible to make these phones subscribe to a mailbox number?  I've done it in the past with Polycoms without any issue but I can't seem to get these to work... 


Thanks for any help!

Spencer



From msc at freeswitch.org  Thu Jun 23 10:19:40 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 22 Jun 2011 23:19:40 -0700
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <AC8EC5A4-1799-433A-9DD0-A715ADF5F256@freeswitch.org>
References: <4E0219B2.9070808@statirasystems.com>
	<4E0227C8.9080304@statirasystems.com>
	<4E02340D.2050708@statirasystems.com>
	<BANLkTiksPsLhd23O-ws626svwnKcJJRmXw@mail.gmail.com>
	<BANLkTinAjxG3Bp7jy3GTRs1km_DdoFDrCA@mail.gmail.com>
	<BANLkTimip1BOfOG+U9rBOAJeiUJK_PL4fw@mail.gmail.com>
	<BANLkTimBC4FKan=kjNoizbGUFGAe1ACu-A@mail.gmail.com>
	<AC8EC5A4-1799-433A-9DD0-A715ADF5F256@freeswitch.org>
Message-ID: <BANLkTikQmkxi_C_1aQocH3O+nQrt4eaA9A@mail.gmail.com>

Perhaps the gateway wants the leading 1 or expects e.164 format?
-MC

On Wed, Jun 22, 2011 at 5:37 PM, Brian West <brian at freeswitch.org> wrote:

> 484 is address incomplete means you didn't send enough digits to the
> gateway usually.
>
> /b
>
> On Jun 22, 2011, at 7:34 PM, Brad Mina wrote:
>
> Ah, thought it was recv not send. Good to know that an invalid gateway
> issue
> can cause this.
>
> On Wed, Jun 22, 2011 at 5:31 PM, Michael Collins <msc at freeswitch.org>
> wrote:
>
> The 484 I believe is a result of the bad gateway. In any case, the OP needs
>
> to figure out what's up w/ that gw or no calls will be going out...
>
>
> -MC
>
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From a.luppi at seletech.com  Thu Jun 23 12:05:43 2011
From: a.luppi at seletech.com (Alessandro)
Date: Thu, 23 Jun 2011 10:05:43 +0200
Subject: [Freeswitch-users] Modified Follow me
In-Reply-To: <BANLkTi=QMN1+MvT-c15FFR8O+r2ERbdBKA@mail.gmail.com>
References: <4E00A61C.1060503@seletech.com>	<BANLkTimSVMrTGJg72JJe5252z7J-RMXSLA@mail.gmail.com>	<4E020A11.4080706@seletech.com>
	<4E024269.8060503@seletech.com>
	<BANLkTi=QMN1+MvT-c15FFR8O+r2ERbdBKA@mail.gmail.com>
Message-ID: <4E02F3D7.4070108@seletech.com>

Hi,

this is the original script created by fusionpbx when enabling follow me 
on an extension:
http://pastebin.freeswitch.org/16565

to solve the problem I have modified the script, removing the answer and 
voice-mail. I also delete the time-out on bridge because i want that the 
second destination is bridged only if the first isn't in the network..

This is the modified script:

http://pastebin.freeswitch.org/16570

best regards

Alessandro

Il 23/06/2011 01:23, Christian Benke ha scritto:
> On 22 June 2011 21:28, Alessandro<a.luppi at seletech.com>  wrote:
>> Solved,
> Could you post the solution to complete this thread please?
> Respectively file a bug-report/feature-improvement-proposal with
> fusionpbx?
>
> Thanks,
> Christian
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


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



From david.ponzone at ipeva.fr  Thu Jun 23 12:21:05 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 23 Jun 2011 10:21:05 +0200
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <BANLkTikQmkxi_C_1aQocH3O+nQrt4eaA9A@mail.gmail.com>
References: <4E0219B2.9070808@statirasystems.com>
	<4E0227C8.9080304@statirasystems.com>
	<4E02340D.2050708@statirasystems.com>
	<BANLkTiksPsLhd23O-ws626svwnKcJJRmXw@mail.gmail.com>
	<BANLkTinAjxG3Bp7jy3GTRs1km_DdoFDrCA@mail.gmail.com>
	<BANLkTimip1BOfOG+U9rBOAJeiUJK_PL4fw@mail.gmail.com>
	<BANLkTimBC4FKan=kjNoizbGUFGAe1ACu-A@mail.gmail.com>
	<AC8EC5A4-1799-433A-9DD0-A715ADF5F256@freeswitch.org>
	<BANLkTikQmkxi_C_1aQocH3O+nQrt4eaA9A@mail.gmail.com>
Message-ID: <B4D334F2-EE02-4EFB-A4A3-165C55CF7606@ipeva.fr>

William,

If I believe the example configuraiton on Junction's web, they don't expect E164:

   <extension name="junctionnetworks.com>
    <condition field="destination_number" expression="^(1{0,1}\d{10})$">
      <action application="set"data="effective_caller_id_number=OUTBOUNDCALLERID"/>
      <action application="bridge" data="sofia/gateway/junctionnetworks.com/$1"/>
     </condition>
   </extension>
But, elsewhere in their knowledgebase, they say:

Numbering
The standard e.164 numbering plan (ITU) is used. North American numbers are required to be prefixed with a '1'. International numbers need to be prefixed with "011".


I think they are confused about what is E164....

You should try to prefix with 1 as Michael recommended.

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

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

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




Le 23/06/2011 ? 08:19, Michael Collins a ?crit :

> Perhaps the gateway wants the leading 1 or expects e.164 format?
> -MC
> 
> On Wed, Jun 22, 2011 at 5:37 PM, Brian West <brian at freeswitch.org> wrote:
> 484 is address incomplete means you didn't send enough digits to the gateway usually.
> 
> /b
> 
> On Jun 22, 2011, at 7:34 PM, Brad Mina wrote:
> 
>> Ah, thought it was recv not send. Good to know that an invalid gateway issue
>> can cause this.
>> 
>> On Wed, Jun 22, 2011 at 5:31 PM, Michael Collins <msc at freeswitch.org> wrote:
>> 
>>> The 484 I believe is a result of the bad gateway. In any case, the OP needs
>>> to figure out what's up w/ that gw or no calls will be going out...
>>> 
>>> -MC
>> 
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at ipeva.fr  Thu Jun 23 12:22:49 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 23 Jun 2011 10:22:49 +0200
Subject: [Freeswitch-users] Minor issue with INVITE domain
In-Reply-To: <201106220751.25728.sos@sokhapkin.dyndns.org>
References: <4BC0A7AA-CBC9-4A52-8C63-9C368F8D04EC@ipeva.fr>
	<201106220751.25728.sos@sokhapkin.dyndns.org>
Message-ID: <2EB0D62C-A1A9-4ECF-B74E-446F950E1FFE@ipeva.fr>

Sergey,

that does not answer to my theoretical question but that indeed solves my issue :)
I was so pissed not to be able to find a way to do that in FS that I disregarded any other possibilities...

Thanks!

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

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

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




Le 22/06/2011 ? 13:51, Sergey Okhapkin a ?crit :

> Put IP address of whatever.foo.com to /etc/hosts and use domain name. you will 
> not get any DNS issues.
> 
> On Wednesday 22 June 2011, David Ponzone wrote:
>> Hello all,
>> 
>> I work with a provider which I need to send calls using their IP (I want to
>> avoid any DNS issues), but I need the FQDN to be in the INVITE domain. So
>> basically, I have a gateway defined with
>> <param name="proxy" value="X.X.X.X"/>
>> but I need the INVITE to be:
>> sip:called at whatever.foo.com
>> and not:
>> sip:called at X.X.X.X
>> 
>> I played with everything I could think of:
>> outbound-proxy in gateway params (which seems to do nothing)
>> 
>> {sip_invite_domain=whatever.foo.com}bridge/gateway/gw/called
>> also does nothing.
>> 
>> I am missing something ?
>> 
>> Thanks
>> 
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> 
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> 
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
>> s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de
>> ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From kees at mroffice.org  Thu Jun 23 13:33:01 2011
From: kees at mroffice.org (Kees Varekamp)
Date: Thu, 23 Jun 2011 21:33:01 +1200
Subject: [Freeswitch-users] mod_rtmp client API
In-Reply-To: <BANLkTikmOfC7ot12-sGcO3aZ7+bH_cyGCQ@mail.gmail.com>
References: <801B6B74CA73484F96B9BBD9BE125561@e1705>
	<BANLkTikmOfC7ot12-sGcO3aZ7+bH_cyGCQ@mail.gmail.com>
Message-ID: <BANLkTik_S5POTuhAOY6YdsEbd9fR8EUGLw@mail.gmail.com>

I'm interested! Can I play around with it?

Kees

On Thu, Jun 23, 2011 at 04:19, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Actually......
>
> We have something we are working on that we are considering releasing
> if we can build a community base effort around it.
> When I get a sign of enough interest we'll go from there.
>
> It would be nice to do some presentations at ClueCon about it.
>
>
> On Wed, Jun 22, 2011 at 10:53 AM, Madovsky <infos at madovsky.org> wrote:
> > if I type "actionscript mod_rtmp freeswitch" on google
> > nothing that helps a common developer to work on mod_rtmp
> > ....
> >
> >
> > ----- Original Message -----
> > From: Madovsky
> > To: FreeSWITCH Users Help
> > Sent: Wednesday, June 22, 2011 11:20 AM
> > Subject: Re: [Freeswitch-users] mod_rtmp client API
> > Mike,
> >
> > I didn't ask tools, but to know at least which function it needs to call
> to
> > the server
> > to make it work ! :)
> >
> > ----- Original Message -----
> > From: Michael Collins
> > To: FreeSWITCH Users Help
> > Sent: Wednesday, June 22, 2011 11:11 AM
> > Subject: Re: [Freeswitch-users] mod_rtmp client API
> > No, it's a DIY project. We provided the server, you provide the client.
> :D
> > Google around and you will find various tools and pieces to start
> building
> > your own.
> > -MC
> >
> > On Wed, Jun 22, 2011 at 7:52 AM, Madovsky <infos at madovsky.org> wrote:
> >>
> >> Is there any client side API to use mod_rtmp ?
> >>
> >> Thanks
> >> _______________________________________________
> >> Join us at ClueCon 2011, Aug 9-11, Chicago
> >> http://www.cluecon.com 877-7-4ACLUE
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> > ________________________________
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From max.asterisk at gmail.com  Thu Jun 23 14:31:15 2011
From: max.asterisk at gmail.com (Max Alex)
Date: Thu, 23 Jun 2011 16:01:15 +0530
Subject: [Freeswitch-users] Answer issue on inbound call
In-Reply-To: <BANLkTim=TOQD1tUvmHjeRuhAR-eKkGYgsw@mail.gmail.com>
References: <BANLkTin+f=uDMiOouyLWAorPvNTd1Z6xaA@mail.gmail.com>
	<BANLkTim=TOQD1tUvmHjeRuhAR-eKkGYgsw@mail.gmail.com>
Message-ID: <BANLkTim3pTPV804t_Usk59eG2C56HAyFdA@mail.gmail.com>

Hi,
Thanks for your reply.
Here is my configuration and log
http://pastebin.freeswitch.org/16571

I am using A200 analog sangoma device with freeswitch, it is working fine
but when it is routing call to 1001 then it is answered.
Please provider your suggestions.

Thanks,
Max Alex
Voip Developer



On Wed, Jun 22, 2011 at 8:51 PM, Michael Collins <msc at freeswitch.org> wrote:

> I thought the A200 was an analog card? Maybe I have my numbers mixed up...
>
> Go ahead and collect a debug log of this call. It might help to have your
> configs posted as well. Use pastebin.freeswitch.org. See this wiki article
> for tips on how to collect information:
> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>
> -MC
>
> On Wed, Jun 22, 2011 at 3:23 AM, Max Alex <max.asterisk at gmail.com> wrote:
>
>> Hi,
>> I have installed freeswitch latest version with sangoma card A200 as well,
>> I have installed and configured freetdm module with wanpipe drivers for
>> freeswitch,
>> We are properly receiving the inbound calls in public context and then we
>> are routing that call to 1001 extension,
>> it is properly routing to 1001 as well, but we have one issue while
>> routing on 1001.
>>
>> Here is the issue description.
>> I am calling from my cell phone to that DID number of pri line, and then
>> it will start ringing on 1001 extension,
>> When 1001 extension start ringing the call is answered on my cell phone,
>> it is something like freeswitch preanswer or autoanswer the call, how can
>> i stop this answer call when it is ringing on 1001 extension,
>> Waiting for good reply.
>>
>> Thanks,
>> Max Alex
>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jaasmailing at gmail.com  Thu Jun 23 16:27:37 2011
From: jaasmailing at gmail.com (Carlo Dimaggio)
Date: Thu, 23 Jun 2011 14:27:37 +0200
Subject: [Freeswitch-users] RTMP/SIP solution - scalability
Message-ID: <4E033139.1040709@gmail.com>

Hi all,

first of all thank you for the mod_rtmp contribution. It is a very 
interesting solution that could be used to create UC platform.
I would like to know if such a this module is suitable in an hosted 
environment (with web portal and flash client) and what could be the 
scalability in this case.
I know it is a new/beta module but I'm wondering if somebody have done 
load test or has thought about this application.

Just another question: when we could have documentation about 
configuration and sip integration?


Best Regards,
Carlo Dimaggio
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From soapee01.fs at stubbornroses.com  Thu Jun 23 18:52:08 2011
From: soapee01.fs at stubbornroses.com (soapee01)
Date: Thu, 23 Jun 2011 09:52:08 -0500
Subject: [Freeswitch-users] FreeTDM - Sangoma B700 - ISDN
 connection	questions - UK
Message-ID: <4E035318.4070807@stubbornroses.com>

On 6/6/2011 4:36 PM, John wrote:

>  Hello,
>
>
>  3. I have built FreeSWITCH from git and installed at /usr/local/... and
>  then followed the steps on the Ubuntu page in the Wiki to set up the run
>  control scripts, etc, and run FS non-root as freeswitch:daemon.  With
>  FreeTDM, I have discovered that the /dev/wan* devices are owned by
>  root:root, and so are inaccessible to FS running as non-root.  So for
>  now I have added a line in /etc/init.d/freeswitch to 'chgrp freeswitch
>  /dev/wan*'.  This is not the most elegant solution, because 'wanrouter
>  restart' (which seems to be my best friend at the moment) resets the
>  ownership to root:root.  I have tried grepping to see where the mknods
>  are for these devices, but have been unsuccessful.  Is there a better
>  place to 'permanently' change the device ownership?
>
John,

Have a look at /etc/udev/rules.d
You can write a rule there and set the ownership permanently that way.

Sorry about the delay on this John. I'd sent it earlier but managed to use
the wrong identity and the list didn't bounce my reply (or I didn't notice).
You've probably figured this out on your own by now...

James



From anthony.minessale at gmail.com  Thu Jun 23 19:42:28 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 23 Jun 2011 10:42:28 -0500
Subject: [Freeswitch-users] second member to dynamic conference hears
 only hold music
In-Reply-To: <BANLkTi=+nT1LT_8pB_7yy_rMf37TFixwcA@mail.gmail.com>
References: <BANLkTikxRYXfOELxVNV9gD6Wx=Y9SN4=vA@mail.gmail.com>
	<BANLkTi=9TWsFmhnqwpom6CDN8U_v5fQYDA@mail.gmail.com>
	<BANLkTi=+nT1LT_8pB_7yy_rMf37TFixwcA@mail.gmail.com>
Message-ID: <BANLkTinQL+jaHU0f87MPHNLhr51K4zB2-Q@mail.gmail.com>

this really should be in jira.
please open one http://jira.freeswitch.org

and post the url here

do the following

1) update right now to the latest GIT as of the time you read this.
2) turn on debugging and sip trace with these commands:
>console loglevel debug
>sofia global siptrace on
3) reproduce your issue and capture the console log
4) attach it to your jira report

Then:

try version 9df8169d1f3458e0b565a64922a1390ebf324703




On Wed, Jun 22, 2011 at 8:00 PM, elijah <elijah at crankenstein.com> wrote:
> That didn't help. Maybe I've screwed this up by running 'git reset --hard'?
> I've confirmed this version works for me:?FreeSWITCH Version 1.0.head
> (git-5923f71 2011-06-01 22-36-19 -0500)
> and this does not:?FreeSWITCH Version 1.0.head (git-288455c 2011-06-22
> 17-05-53 -0400)
> On Wed, Jun 22, 2011 at 5:30 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> could you comment line 1696 of switch_channel.c and
>> make install_core
>> then restart and retest?
>>
>> That is the only one I can think of.
>>
>> On Wed, Jun 22, 2011 at 5:53 PM, elijah <elijah at crankenstein.com> wrote:
>> > This setup was working for me until updating against git yesterday. Via
>> > a
>> > bind statement on the b-leg of inbound calls, a user is able to pull
>> > both
>> > legs into a conference bridge. The problem now is that the a-leg hears
>> > only
>> > hold music. I'm having this and another problem with my dialplan after
>> > updating, w/o changing my own configuration - could there be updates to
>> > source that would be affecting me? Here's the setup:
>> > ? ? <extension name="telifi_sales_queue">
>> > ? ? ? <condition field="destination_number"
>> > expression="^telifi_sales_queue$">
>> > ? ? ? ? <action application="bridge_export"
>> > data="telifi_call_id=${telifi_call_id}"/>
>> > ? ? ? ? <action application="set" data="bind_meta_key=#"/>
>> > ? ? ? ? <action application="bind_meta_app" data="1 b s
>> > execute_extension::dx1 XML callsdirect"/>
>> > ? ? ? ? <action application="bind_meta_app" data="2 b s
>> > execute_extension::dx2 XML callsdirect"/>
>> > ? ? ? ? <action application="callcenter"
>> > data="${telifi_queue_name}@live001.voice.telifi.com"/>
>> > ? ? ? </condition>
>> > ? ? </extension>
>> > ? ? <extension name="bind_receiver_2">
>> > ? ? ? <condition field="destination_number" expression="^dx2$">
>> > ? ? ? ? <action application="answer"/>
>> > ? ? ? ? <action application="transfer" data="-both dynamic_conference
>> > XML
>> > callsdirect"/>
>> > ? ? ? </condition>
>> > ? ? </extension>
>> > ? ? <extension name="dynamic_conference">
>> > ? ? ? <condition field="destination_number"
>> > expression="^dynamic_conference"/>
>> > ? ? ? <condition field="rdnis" expression="^(dx2)$">
>> > ? ? ? ? <action application="answer"/>
>> > ? ? ? ? <action application="bind_digit_action"
>> >
>> > data="moderator,*1,exec:execute_extension,ASK_FOR_NUMBER__${telifi_call_id}
>> > XML callsdirect"/>
>> > ? ? ? ? <action application="bind_digit_action"
>> >
>> > data="moderator,*2,exec:execute_extension,CANCEL_LAST_CALL__${telifi_call_id}
>> > XML callsdirect"/>
>> > ? ? ? ? <action application="digit_action_set_realm" data="moderator"/>
>> > ? ? ? ? <action application="conference"
>> > data="${telifi_call_id}@simple"/>
>> > ? ? ? ? <anti-action application="answer"/>
>> > ? ? ? ? <anti-action application="conference"
>> > data="${telifi_call_id}@simple"/>
>> > ? ? ? </condition>
>> > ? ? </extension>
>> >
>> > _______________________________________________
>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>> > http://www.cluecon.com 877-7-4ACLUE
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From zahid.maqbool at gmail.com  Thu Jun 23 19:49:16 2011
From: zahid.maqbool at gmail.com (Zahid Maqbool)
Date: Thu, 23 Jun 2011 16:49:16 +0100
Subject: [Freeswitch-users] Off Hook Agents in Mod_Fifo
Message-ID: <BANLkTinT5YzRFKJEb9mJw_s=oW_jR1FpGw@mail.gmail.com>

Hi,

I've been trying out Freeswitch and so far it seems great, however I wanted
to test out call center features. So I had a look at Mod_FIFO. I'd like to
understand how to configure off hook agents and if its possible to login an
off hook agent via an external application? Looking at the mod FIFO most of
the examples are for on hook agents, does anyone know how to configure off
hook agents?

What exactly I want to do is let's say, there is an agent application on the
desktop and I want the agent to put his phone off hook, enter the agent id
and extension through this application and click on login.

Any pointers would be appreciated.

Thanks.


Zahid Maqbool
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From anthony.minessale at gmail.com  Thu Jun 23 20:02:30 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 23 Jun 2011 11:02:30 -0500
Subject: [Freeswitch-users] Off Hook Agents in Mod_Fifo
In-Reply-To: <BANLkTinT5YzRFKJEb9mJw_s=oW_jR1FpGw@mail.gmail.com>
References: <BANLkTinT5YzRFKJEb9mJw_s=oW_jR1FpGw@mail.gmail.com>
Message-ID: <BANLkTim-iR0ZWipAbDBAyZ5550258MTxjg@mail.gmail.com>

if you enter the fifo app as an agent and supply the wait keyword in
place of nowait, the agent remains off hook until a call is received.


On Thu, Jun 23, 2011 at 10:49 AM, Zahid Maqbool <zahid.maqbool at gmail.com> wrote:
> Hi,
>
> I've been trying out Freeswitch and so far it seems great, however I wanted
> to test out call center features. So I had a look at Mod_FIFO. I'd like to
> understand how to configure off hook agents and if its possible to login an
> off hook agent via an external application? Looking at the mod FIFO most of
> the examples are for on hook agents, does anyone know how to configure off
> hook agents?
>
> What exactly I want to do is let's say, there is an agent application on the
> desktop and I want the agent to put his phone off hook, enter the agent id
> and extension through this application and click on login.
>
> Any pointers would be appreciated.
>
> Thanks.
>
>
> Zahid Maqbool
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From bmoore at statirasystems.com  Thu Jun 23 20:15:24 2011
From: bmoore at statirasystems.com (William Moore)
Date: Thu, 23 Jun 2011 12:15:24 -0400
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <B4D334F2-EE02-4EFB-A4A3-165C55CF7606@ipeva.fr>
References: <4E0219B2.9070808@statirasystems.com>	<4E0227C8.9080304@statirasystems.com>	<4E02340D.2050708@statirasystems.com>	<BANLkTiksPsLhd23O-ws626svwnKcJJRmXw@mail.gmail.com>	<BANLkTinAjxG3Bp7jy3GTRs1km_DdoFDrCA@mail.gmail.com>	<BANLkTimip1BOfOG+U9rBOAJeiUJK_PL4fw@mail.gmail.com>	<BANLkTimBC4FKan=kjNoizbGUFGAe1ACu-A@mail.gmail.com>	<AC8EC5A4-1799-433A-9DD0-A715ADF5F256@freeswitch.org>	<BANLkTikQmkxi_C_1aQocH3O+nQrt4eaA9A@mail.gmail.com>
	<B4D334F2-EE02-4EFB-A4A3-165C55CF7606@ipeva.fr>
Message-ID: <4E03669C.2030603@statirasystems.com>

It was the 1 and it needs to be set for e.164(+1). Blue.box makes it a 
little more difficult since it wants to manage the xmls. I could have 
set it up as a separate config file however that defeats the purpose of 
the GUI.

I still have the issue of incoming calls not even showing up in sofia. 
Junction Networks says it is going out, but my server is not responding. 
I'm not sure if I have the settings for incoming correct.

William J. Moore
Statira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com


On 06/23/2011 04:21 AM, David Ponzone wrote:
> William,
>
> If I believe the example configuraiton on Junction's web, they don't 
> expect E164:
>
>     <extension name="junctionnetworks.com  <http://junctionnetworks.com>>
>      <condition field="destination_number" expression="^(1{0,1}\d{10})$">
>        <action application="set"data="effective_caller_id_number=OUTBOUNDCALLERID"/>
>        <action application="bridge" data="sofia/gateway/junctionnetworks.com/$1"/>
>       </condition>
>     </extension>
> But, elsewhere in their knowledgebase, they say:
>
> *Numbering*
> The standard e.164 numbering plan (ITU <http://www.itu.int/>) is used. 
> North American numbers are required to be prefixed with a '1'. 
> International numbers need to be prefixed with "011".
>
>
> I think they are confused about what is E164....
>
> You should try to prefix with 1 as Michael recommended.
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service ClientIPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
> diffusion non autoris?e est interdite. Tout message ?lectronique est 
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au 
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
> n'?tes pas destinataire de ce message, merci de le d?truire 
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 23/06/2011 ? 08:19, Michael Collins a ?crit :
>
>> Perhaps the gateway wants the leading 1 or expects e.164 format?
>> -MC
>>
>> On Wed, Jun 22, 2011 at 5:37 PM, Brian West <brian at freeswitch.org 
>> <mailto:brian at freeswitch.org>> wrote:
>>
>>     484 is address incomplete means you didn't send enough digits to
>>     the gateway usually.
>>
>>     /b
>>
>>     On Jun 22, 2011, at 7:34 PM, Brad Mina wrote:
>>
>>>     Ah, thought it was recv not send. Good to know that an invalid
>>>     gateway issue
>>>     can cause this.
>>>
>>>     On Wed, Jun 22, 2011 at 5:31 PM, Michael Collins
>>>     <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
>>>
>>>>     The 484 I believe is a result of the bad gateway. In any case,
>>>>     the OP needs
>>>>     to figure out what's up w/ that gw or no calls will be going out...
>>>>
>>>>     -MC
>>>
>>
>>
>>     _______________________________________________
>>     Join us at ClueCon 2011, Aug 9-11, Chicago
>>     http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org 
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From jack at livecall.com  Thu Jun 23 19:48:11 2011
From: jack at livecall.com (Jack)
Date: Thu, 23 Jun 2011 08:48:11 -0700
Subject: [Freeswitch-users] mod_rtmp client API
In-Reply-To: <BANLkTikmOfC7ot12-sGcO3aZ7+bH_cyGCQ@mail.gmail.com>
References: <801B6B74CA73484F96B9BBD9BE125561@e1705>
	<BANLkTikmOfC7ot12-sGcO3aZ7+bH_cyGCQ@mail.gmail.com>
Message-ID: <4E03603B.6020405@livecall.com>

I am interested....  It looks like a clean, simple way to connect to 
Freeswitch.
Nice Job!
Jack

On 6/22/2011 9:19 AM, Anthony Minessale wrote:
> Actually......
>
> We have something we are working on that we are considering releasing
> if we can build a community base effort around it.
> When I get a sign of enough interest we'll go from there.
>
> It would be nice to do some presentations at ClueCon about it.
>
>
> On Wed, Jun 22, 2011 at 10:53 AM, Madovsky<infos at madovsky.org>  wrote:
>> if I type "actionscript mod_rtmp freeswitch" on google
>> nothing that helps a common developer to work on mod_rtmp
>> ....
>>
>>
>> ----- Original Message -----
>> From: Madovsky
>> To: FreeSWITCH Users Help
>> Sent: Wednesday, June 22, 2011 11:20 AM
>> Subject: Re: [Freeswitch-users] mod_rtmp client API
>> Mike,
>>
>> I didn't ask tools, but to know at least which function it needs to call to
>> the server
>> to make it work ! :)
>>
>> ----- Original Message -----
>> From: Michael Collins
>> To: FreeSWITCH Users Help
>> Sent: Wednesday, June 22, 2011 11:11 AM
>> Subject: Re: [Freeswitch-users] mod_rtmp client API
>> No, it's a DIY project. We provided the server, you provide the client. :D
>> Google around and you will find various tools and pieces to start building
>> your own.
>> -MC
>>
>> On Wed, Jun 22, 2011 at 7:52 AM, Madovsky<infos at madovsky.org>  wrote:
>>> Is there any client side API to use mod_rtmp ?
>>>
>>> Thanks
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>> ________________________________
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>


From mik3weider at gmail.com  Thu Jun 23 20:20:48 2011
From: mik3weider at gmail.com (=?KOI8-R?B?88zA08HSxdcg7cnIwcnM?=)
Date: Thu, 23 Jun 2011 20:20:48 +0400
Subject: [Freeswitch-users] FreeSWITCHer: issue with ObjectSpace
Message-ID: <BANLkTik7cz8OoNxL7di5-X1+2-s9pn28dA@mail.gmail.com>

Hey, guys

My question is about FreeSWITCHer the Ruby framework handle events
from FreeSWITCH. I know that this is not exactly what you are discuss
here, but I really don't know whith whom to comunicate about this
problem. So, if you have experience in FreeSWITCHer and Ruby and would
like to help, please read futher. Otherwise, sorry for spam.

FreeSWITCHer doesn't sent subscribing messages during of usage of
examples/inbound_socket_events.rb. As I can see in the source, that
happens because subscribe_to_event function is never trigered.

So, the question is why this snippet of code doesn't work as expected?

def self.add_event_hook(event, sub_events = [], &block)
        ObjectSpace.each_object { |e| e.subscribe_to_event(event,
sub_events) if e.class.ancestors.include?(FSR::Listener::Inbound) }
        HOOKS[event] = block
end

And this is my environment:

# cat /etc/redhat-release
CentOS release 5.6 (Final)

# ruby -v
ruby 1.8.5 (2006-08-25) [i386-linux]

# gem -v
1.3.5

# gem list

*** LOCAL GEMS ***

eventmachine (0.12.10)
freeswitcher (0.6.12)
log4r (1.1.9)

Since it's looks like Ruby configuration issue I don't describe other
installation steps.


From david.ponzone at ipeva.fr  Thu Jun 23 20:25:14 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 23 Jun 2011 18:25:14 +0200
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <4E03669C.2030603@statirasystems.com>
References: <4E0219B2.9070808@statirasystems.com>	<4E0227C8.9080304@statirasystems.com>	<4E02340D.2050708@statirasystems.com>	<BANLkTiksPsLhd23O-ws626svwnKcJJRmXw@mail.gmail.com>	<BANLkTinAjxG3Bp7jy3GTRs1km_DdoFDrCA@mail.gmail.com>	<BANLkTimip1BOfOG+U9rBOAJeiUJK_PL4fw@mail.gmail.com>	<BANLkTimBC4FKan=kjNoizbGUFGAe1ACu-A@mail.gmail.com>	<AC8EC5A4-1799-433A-9DD0-A715ADF5F256@freeswitch.org>	<BANLkTikQmkxi_C_1aQocH3O+nQrt4eaA9A@mail.gmail.com>
	<B4D334F2-EE02-4EFB-A4A3-165C55CF7606@ipeva.fr>
	<4E03669C.2030603@statirasystems.com>
Message-ID: <963C184F-7466-4874-80F1-0CC13FEC8581@ipeva.fr>

Take a sip trace on your box with tcpdump or preferably, tshark:
tshark port 5080

You will then see if you receive it.

In the config you sent, I don't see anything about register expiration, or NAT keepalive, so I would really recommend you add a ping every 30 seconds to your gateway config.

If the issue is there, it's quite easy to see:
unregister the gateway
register it again
make an inbound call right away

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

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

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




Le 23/06/2011 ? 18:15, William Moore a ?crit :

> It was the 1 and it needs to be set for e.164(+1). Blue.box makes it a little more difficult since it wants to manage the xmls. I could have set it up as a separate config file however that defeats the purpose of the GUI.
> 
> I still have the issue of incoming calls not even showing up in sofia. Junction Networks says it is going out, but my server is not responding. I'm not sure if I have the settings for incoming correct.
> William J. Moore
> Statira Systems
> 611 Caroline St
> Fredericksburg, VA
> http://www.statirasystems.com
> 
> On 06/23/2011 04:21 AM, David Ponzone wrote:
>> 
>> William,
>> 
>> If I believe the example configuraiton on Junction's web, they don't expect E164:
>> 
>>     <extension name="junctionnetworks.com>
>>     <condition field="destination_number" expression="^(1{0,1}\d{10})$">
>>       <action application="set"data="effective_caller_id_number=OUTBOUNDCALLERID"/>
>>       <action application="bridge" data="sofia/gateway/junctionnetworks.com/$1"/>
>>      </condition>
>>    </extension>
>> But, elsewhere in their knowledgebase, they say:
>> 
>> Numbering
>> The standard e.164 numbering plan (ITU) is used. North American numbers are required to be prefixed with a '1'. International numbers need to be prefixed with "011".
>> 
>> 
>> I think they are confused about what is E164....
>> 
>> You should try to prefix with 1 as Michael recommended.
>> 
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> 
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> 
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>> 
>> 
>> 
>> 
>> Le 23/06/2011 ? 08:19, Michael Collins a ?crit :
>> 
>>> Perhaps the gateway wants the leading 1 or expects e.164 format?
>>> -MC
>>> 
>>> On Wed, Jun 22, 2011 at 5:37 PM, Brian West <brian at freeswitch.org> wrote:
>>> 484 is address incomplete means you didn't send enough digits to the gateway usually.
>>> 
>>> /b
>>> 
>>> On Jun 22, 2011, at 7:34 PM, Brad Mina wrote:
>>> 
>>>> Ah, thought it was recv not send. Good to know that an invalid gateway issue
>>>> can cause this.
>>>> 
>>>> On Wed, Jun 22, 2011 at 5:31 PM, Michael Collins <msc at freeswitch.org> wrote:
>>>> 
>>>>> The 484 I believe is a result of the bad gateway. In any case, the OP needs
>>>>> to figure out what's up w/ that gw or no calls will be going out...
>>>>> 
>>>>> -MC
>>>> 
>>> 
>>> 
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From zahid.maqbool at gmail.com  Thu Jun 23 20:25:30 2011
From: zahid.maqbool at gmail.com (Zahid Maqbool)
Date: Thu, 23 Jun 2011 17:25:30 +0100
Subject: [Freeswitch-users] Off Hook Agents in Mod_Fifo
In-Reply-To: <BANLkTim-iR0ZWipAbDBAyZ5550258MTxjg@mail.gmail.com>
References: <BANLkTinT5YzRFKJEb9mJw_s=oW_jR1FpGw@mail.gmail.com>
	<BANLkTim-iR0ZWipAbDBAyZ5550258MTxjg@mail.gmail.com>
Message-ID: <BANLkTimCoqY+eVg-c0J2jkyHd5FZOD_dbg@mail.gmail.com>

Thanks a lot Anthony.

On Thu, Jun 23, 2011 at 5:02 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> if you enter the fifo app as an agent and supply the wait keyword in
> place of nowait, the agent remains off hook until a call is received.
>
>
> On Thu, Jun 23, 2011 at 10:49 AM, Zahid Maqbool <zahid.maqbool at gmail.com>
> wrote:
> > Hi,
> >
> > I've been trying out Freeswitch and so far it seems great, however I
> wanted
> > to test out call center features. So I had a look at Mod_FIFO. I'd like
> to
> > understand how to configure off hook agents and if its possible to login
> an
> > off hook agent via an external application? Looking at the mod FIFO most
> of
> > the examples are for on hook agents, does anyone know how to configure
> off
> > hook agents?
> >
> > What exactly I want to do is let's say, there is an agent application on
> the
> > desktop and I want the agent to put his phone off hook, enter the agent
> id
> > and extension through this application and click on login.
> >
> > Any pointers would be appreciated.
> >
> > Thanks.
> >
> >
> > Zahid Maqbool
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kind Regards,

Zahid Maqbool
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From bmoore at statirasystems.com  Thu Jun 23 20:50:45 2011
From: bmoore at statirasystems.com (William Moore)
Date: Thu, 23 Jun 2011 12:50:45 -0400
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <963C184F-7466-4874-80F1-0CC13FEC8581@ipeva.fr>
References: <4E0219B2.9070808@statirasystems.com>	<4E0227C8.9080304@statirasystems.com>	<4E02340D.2050708@statirasystems.com>	<BANLkTiksPsLhd23O-ws626svwnKcJJRmXw@mail.gmail.com>	<BANLkTinAjxG3Bp7jy3GTRs1km_DdoFDrCA@mail.gmail.com>	<BANLkTimip1BOfOG+U9rBOAJeiUJK_PL4fw@mail.gmail.com>	<BANLkTimBC4FKan=kjNoizbGUFGAe1ACu-A@mail.gmail.com>	<AC8EC5A4-1799-433A-9DD0-A715ADF5F256@freeswitch.org>	<BANLkTikQmkxi_C_1aQocH3O+nQrt4eaA9A@mail.gmail.com>	<B4D334F2-EE02-4EFB-A4A3-165C55CF7606@ipeva.fr>	<4E03669C.2030603@statirasystems.com>
	<963C184F-7466-4874-80F1-0CC13FEC8581@ipeva.fr>
Message-ID: <4E036EE5.60208@statirasystems.com>

I get this with wireshark

  40.345679 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
  40.347381 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy 
Authentication Required
  40.367906 66.227.100.20 -> 192.168.1.254 SIP Request: ACK 
sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2

I am looking for the expiration now..

William J. Moore
Statira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com


On 06/23/2011 12:25 PM, David Ponzone wrote:
> Take a sip trace on your box with tcpdump or preferably, tshark:
> tshark port 5080
>
> You will then see if you receive it.
>
> In the config you sent, I don't see anything about register 
> expiration, or NAT keepalive, so I would really recommend you add a 
> ping every 30 seconds to your gateway config.
>
> If the issue is there, it's quite easy to see:
> unregister the gateway
> register it again
> make an inbound call right away
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service ClientIPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
> diffusion non autoris?e est interdite. Tout message ?lectronique est 
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au 
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
> n'?tes pas destinataire de ce message, merci de le d?truire 
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 23/06/2011 ? 18:15, William Moore a ?crit :
>
>> It was the 1 and it needs to be set for e.164(+1). Blue.box makes it 
>> a little more difficult since it wants to manage the xmls. I could 
>> have set it up as a separate config file however that defeats the 
>> purpose of the GUI.
>>
>> I still have the issue of incoming calls not even showing up in 
>> sofia. Junction Networks says it is going out, but my server is not 
>> responding. I'm not sure if I have the settings for incoming correct.
>> William J. Moore
>> Statira Systems
>> 611 Caroline St
>> Fredericksburg, VA
>> http://www.statirasystems.com
>>
>> On 06/23/2011 04:21 AM, David Ponzone wrote:
>>> William,
>>>
>>> If I believe the example configuraiton on Junction's web, they don't 
>>> expect E164:
>>>
>>>     <extension name="junctionnetworks.com  <http://junctionnetworks.com/>>
>>>      <condition field="destination_number" expression="^(1{0,1}\d{10})$">
>>>        <action application="set"data="effective_caller_id_number=OUTBOUNDCALLERID"/>
>>>        <action application="bridge" data="sofia/gateway/junctionnetworks.com/$1"/>
>>>       </condition>
>>>     </extension>
>>> But, elsewhere in their knowledgebase, they say:
>>>
>>> *Numbering*
>>> The standard e.164 numbering plan (ITU <http://www.itu.int/>) is 
>>> used. North American numbers are required to be prefixed with a '1'. 
>>> International numbers need to be prefixed with "011".
>>>
>>>
>>> I think they are confused about what is E164....
>>>
>>> You should try to prefix with 1 as Michael recommended.
>>>
>>> David Ponzone Direction Technique
>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service ClientIPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
>>> <BLOCKED::http://www.ipeva-studio.com/>
>>>
>>> /Ce message et toutes les pi?ces jointes sont confidentiels et 
>>> ?tablis ? l'intention exclusive de ses destinataires. Toute 
>>> utilisation ou diffusion non autoris?e est interdite. Tout message 
>>> ?lectronique est susceptible d'alt?ration. /*/IPeva/*/ d?cline toute 
>>> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou 
>>> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le 
>>> d?truire imm?diatement et d'avertir l'exp?diteur./
>>> /
>>> /
>>>
>>>
>>>
>>> Le 23/06/2011 ? 08:19, Michael Collins a ?crit :
>>>
>>>> Perhaps the gateway wants the leading 1 or expects e.164 format?
>>>> -MC
>>>>
>>>> On Wed, Jun 22, 2011 at 5:37 PM, Brian West <brian at freeswitch.org 
>>>> <mailto:brian at freeswitch.org>> wrote:
>>>>
>>>>     484 is address incomplete means you didn't send enough digits
>>>>     to the gateway usually.
>>>>
>>>>     /b
>>>>
>>>>     On Jun 22, 2011, at 7:34 PM, Brad Mina wrote:
>>>>
>>>>>     Ah, thought it was recv not send. Good to know that an invalid
>>>>>     gateway issue
>>>>>     can cause this.
>>>>>
>>>>>     On Wed, Jun 22, 2011 at 5:31 PM, Michael Collins
>>>>>     <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
>>>>>
>>>>>>     The 484 I believe is a result of the bad gateway. In any
>>>>>>     case, the OP needs
>>>>>>     to figure out what's up w/ that gw or no calls will be going
>>>>>>     out...
>>>>>>
>>>>>>     -MC
>>>>>
>>>>
>>>>
>>>>     _______________________________________________
>>>>     Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>     http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>>
>>>>     FreeSWITCH-users mailing list
>>>>     FreeSWITCH-users at lists.freeswitch.org
>>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org 
>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com  877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org 
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From bmoore at statirasystems.com  Thu Jun 23 20:52:11 2011
From: bmoore at statirasystems.com (William Moore)
Date: Thu, 23 Jun 2011 12:52:11 -0400
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <4E036EE5.60208@statirasystems.com>
References: <4E0219B2.9070808@statirasystems.com>	<4E0227C8.9080304@statirasystems.com>	<4E02340D.2050708@statirasystems.com>	<BANLkTiksPsLhd23O-ws626svwnKcJJRmXw@mail.gmail.com>	<BANLkTinAjxG3Bp7jy3GTRs1km_DdoFDrCA@mail.gmail.com>	<BANLkTimip1BOfOG+U9rBOAJeiUJK_PL4fw@mail.gmail.com>	<BANLkTimBC4FKan=kjNoizbGUFGAe1ACu-A@mail.gmail.com>	<AC8EC5A4-1799-433A-9DD0-A715ADF5F256@freeswitch.org>	<BANLkTikQmkxi_C_1aQocH3O+nQrt4eaA9A@mail.gmail.com>	<B4D334F2-EE02-4EFB-A4A3-165C55CF7606@ipeva.fr>	<4E03669C.2030603@statirasystems.com>
	<963C184F-7466-4874-80F1-0CC13FEC8581@ipeva.fr>
	<4E036EE5.60208@statirasystems.com>
Message-ID: <4E036F3B.8090303@statirasystems.com>

correction: missed a line.

   0.000000 66.227.100.20 -> 192.168.1.254 SIP/SDP Request: INVITE 
sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with 
session description
   0.000814 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
   0.002440 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy 
Authentication Required
   0.023173 66.227.100.20 -> 192.168.1.254 SIP Request: ACK 
sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2

William J. Moore
Statira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com


On 06/23/2011 12:50 PM, William Moore wrote:
> I get this with wireshark
>
>  40.345679 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>  40.347381 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy 
> Authentication Required
>  40.367906 66.227.100.20 -> 192.168.1.254 SIP Request: ACK 
> sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>
> I am looking for the expiration now..
> William J. Moore
> Statira Systems
> 611 Caroline St
> Fredericksburg, VA
> http://www.statirasystems.com
>
> On 06/23/2011 12:25 PM, David Ponzone wrote:
>> Take a sip trace on your box with tcpdump or preferably, tshark:
>> tshark port 5080
>>
>> You will then see if you receive it.
>>
>> In the config you sent, I don't see anything about register 
>> expiration, or NAT keepalive, so I would really recommend you add a 
>> ping every 30 seconds to your gateway config.
>>
>> If the issue is there, it's quite easy to see:
>> unregister the gateway
>> register it again
>> make an inbound call right away
>>
>> David Ponzone Direction Technique
>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service ClientIPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
>> <BLOCKED::http://www.ipeva-studio.com/>
>>
>> /Ce message et toutes les pi?ces jointes sont confidentiels et 
>> ?tablis ? l'intention exclusive de ses destinataires. Toute 
>> utilisation ou diffusion non autoris?e est interdite. Tout message 
>> ?lectronique est susceptible d'alt?ration. /*/IPeva/*/ d?cline toute 
>> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou 
>> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le 
>> d?truire imm?diatement et d'avertir l'exp?diteur./
>> /
>> /
>>
>>
>>
>> Le 23/06/2011 ? 18:15, William Moore a ?crit :
>>
>>> It was the 1 and it needs to be set for e.164(+1). Blue.box makes it 
>>> a little more difficult since it wants to manage the xmls. I could 
>>> have set it up as a separate config file however that defeats the 
>>> purpose of the GUI.
>>>
>>> I still have the issue of incoming calls not even showing up in 
>>> sofia. Junction Networks says it is going out, but my server is not 
>>> responding. I'm not sure if I have the settings for incoming correct.
>>> William J. Moore
>>> Statira Systems
>>> 611 Caroline St
>>> Fredericksburg, VA
>>> http://www.statirasystems.com
>>>
>>> On 06/23/2011 04:21 AM, David Ponzone wrote:
>>>> William,
>>>>
>>>> If I believe the example configuraiton on Junction's web, they 
>>>> don't expect E164:
>>>>
>>>>     <extension name="junctionnetworks.com  <http://junctionnetworks.com/>>
>>>>      <condition field="destination_number" expression="^(1{0,1}\d{10})$">
>>>>        <action application="set"data="effective_caller_id_number=OUTBOUNDCALLERID"/>
>>>>        <action application="bridge" data="sofia/gateway/junctionnetworks.com/$1"/>
>>>>       </condition>
>>>>     </extension>
>>>> But, elsewhere in their knowledgebase, they say:
>>>>
>>>> *Numbering*
>>>> The standard e.164 numbering plan (ITU <http://www.itu.int/>) is 
>>>> used. North American numbers are required to be prefixed with a 
>>>> '1'. International numbers need to be prefixed with "011".
>>>>
>>>>
>>>> I think they are confused about what is E164....
>>>>
>>>> You should try to prefix with 1 as Michael recommended.
>>>>
>>>> David Ponzone Direction Technique
>>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>>> tel:      01 74 03 18 97
>>>> gsm:   06 66 98 76 34
>>>>
>>>> Service ClientIPeva
>>>> tel:      0811 46 26 26
>>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - 
>>>> www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/>
>>>>
>>>> /Ce message et toutes les pi?ces jointes sont confidentiels et 
>>>> ?tablis ? l'intention exclusive de ses destinataires. Toute 
>>>> utilisation ou diffusion non autoris?e est interdite. Tout message 
>>>> ?lectronique est susceptible d'alt?ration. /*/IPeva/*/ d?cline 
>>>> toute responsabilit? au titre de ce message s'il a ?t? alt?r?, 
>>>> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, 
>>>> merci de le d?truire imm?diatement et d'avertir l'exp?diteur./
>>>> /
>>>> /
>>>>
>>>>
>>>>
>>>> Le 23/06/2011 ? 08:19, Michael Collins a ?crit :
>>>>
>>>>> Perhaps the gateway wants the leading 1 or expects e.164 format?
>>>>> -MC
>>>>>
>>>>> On Wed, Jun 22, 2011 at 5:37 PM, Brian West <brian at freeswitch.org 
>>>>> <mailto:brian at freeswitch.org>> wrote:
>>>>>
>>>>>     484 is address incomplete means you didn't send enough digits
>>>>>     to the gateway usually.
>>>>>
>>>>>     /b
>>>>>
>>>>>     On Jun 22, 2011, at 7:34 PM, Brad Mina wrote:
>>>>>
>>>>>>     Ah, thought it was recv not send. Good to know that an
>>>>>>     invalid gateway issue
>>>>>>     can cause this.
>>>>>>
>>>>>>     On Wed, Jun 22, 2011 at 5:31 PM, Michael Collins
>>>>>>     <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
>>>>>>
>>>>>>>     The 484 I believe is a result of the bad gateway. In any
>>>>>>>     case, the OP needs
>>>>>>>     to figure out what's up w/ that gw or no calls will be going
>>>>>>>     out...
>>>>>>>
>>>>>>>     -MC
>>>>>>
>>>>>
>>>>>
>>>>>     _______________________________________________
>>>>>     Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>     http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>>>
>>>>>     FreeSWITCH-users mailing list
>>>>>     FreeSWITCH-users at lists.freeswitch.org
>>>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org 
>>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com  877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org 
>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com  877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
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From bmoore at statirasystems.com  Thu Jun 23 20:58:01 2011
From: bmoore at statirasystems.com (William Moore)
Date: Thu, 23 Jun 2011 12:58:01 -0400
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <4E036F3B.8090303@statirasystems.com>
References: <4E0219B2.9070808@statirasystems.com>	<4E0227C8.9080304@statirasystems.com>	<4E02340D.2050708@statirasystems.com>	<BANLkTiksPsLhd23O-ws626svwnKcJJRmXw@mail.gmail.com>	<BANLkTinAjxG3Bp7jy3GTRs1km_DdoFDrCA@mail.gmail.com>	<BANLkTimip1BOfOG+U9rBOAJeiUJK_PL4fw@mail.gmail.com>	<BANLkTimBC4FKan=kjNoizbGUFGAe1ACu-A@mail.gmail.com>	<AC8EC5A4-1799-433A-9DD0-A715ADF5F256@freeswitch.org>	<BANLkTikQmkxi_C_1aQocH3O+nQrt4eaA9A@mail.gmail.com>	<B4D334F2-EE02-4EFB-A4A3-165C55CF7606@ipeva.fr>	<4E03669C.2030603@statirasystems.com>
	<963C184F-7466-4874-80F1-0CC13FEC8581@ipeva.fr>
	<4E036EE5.60208@statirasystems.com>
	<4E036F3B.8090303@statirasystems.com>
Message-ID: <4E037099.9090107@statirasystems.com>

When I put it in and it reloads, blue.box removes it. Can I reference 
the gateway in another sip interface file or does it have to be in that one?

William J. Moore
Statira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com


On 06/23/2011 12:52 PM, William Moore wrote:
> correction: missed a line.
>
>   0.000000 66.227.100.20 -> 192.168.1.254 SIP/SDP Request: INVITE 
> sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with 
> session description
>   0.000814 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>   0.002440 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy 
> Authentication Required
>   0.023173 66.227.100.20 -> 192.168.1.254 SIP Request: ACK 
> sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>
> William J. Moore
> Statira Systems
> 611 Caroline St
> Fredericksburg, VA
> http://www.statirasystems.com
>
> On 06/23/2011 12:50 PM, William Moore wrote:
>> I get this with wireshark
>>
>>  40.345679 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>>  40.347381 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy 
>> Authentication Required
>>  40.367906 66.227.100.20 -> 192.168.1.254 SIP Request: ACK 
>> sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>
>> I am looking for the expiration now..
>> William J. Moore
>> Statira Systems
>> 611 Caroline St
>> Fredericksburg, VA
>> http://www.statirasystems.com
>>
>> On 06/23/2011 12:25 PM, David Ponzone wrote:
>>> Take a sip trace on your box with tcpdump or preferably, tshark:
>>> tshark port 5080
>>>
>>> You will then see if you receive it.
>>>
>>> In the config you sent, I don't see anything about register 
>>> expiration, or NAT keepalive, so I would really recommend you add a 
>>> ping every 30 seconds to your gateway config.
>>>
>>> If the issue is there, it's quite easy to see:
>>> unregister the gateway
>>> register it again
>>> make an inbound call right away
>>>
>>> David Ponzone Direction Technique
>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service ClientIPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
>>> <BLOCKED::http://www.ipeva-studio.com/>
>>>
>>> /Ce message et toutes les pi?ces jointes sont confidentiels et 
>>> ?tablis ? l'intention exclusive de ses destinataires. Toute 
>>> utilisation ou diffusion non autoris?e est interdite. Tout message 
>>> ?lectronique est susceptible d'alt?ration. /*/IPeva/*/ d?cline toute 
>>> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou 
>>> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le 
>>> d?truire imm?diatement et d'avertir l'exp?diteur./
>>> /
>>> /
>>>
>>>
>>>
>>> Le 23/06/2011 ? 18:15, William Moore a ?crit :
>>>
>>>> It was the 1 and it needs to be set for e.164(+1). Blue.box makes 
>>>> it a little more difficult since it wants to manage the xmls. I 
>>>> could have set it up as a separate config file however that defeats 
>>>> the purpose of the GUI.
>>>>
>>>> I still have the issue of incoming calls not even showing up in 
>>>> sofia. Junction Networks says it is going out, but my server is not 
>>>> responding. I'm not sure if I have the settings for incoming correct.
>>>> William J. Moore
>>>> Statira Systems
>>>> 611 Caroline St
>>>> Fredericksburg, VA
>>>> http://www.statirasystems.com
>>>>
>>>> On 06/23/2011 04:21 AM, David Ponzone wrote:
>>>>> William,
>>>>>
>>>>> If I believe the example configuraiton on Junction's web, they 
>>>>> don't expect E164:
>>>>>
>>>>>     <extension name="junctionnetworks.com  <http://junctionnetworks.com/>>
>>>>>      <condition field="destination_number" expression="^(1{0,1}\d{10})$">
>>>>>        <action application="set"data="effective_caller_id_number=OUTBOUNDCALLERID"/>
>>>>>        <action application="bridge" data="sofia/gateway/junctionnetworks.com/$1"/>
>>>>>       </condition>
>>>>>     </extension>
>>>>> But, elsewhere in their knowledgebase, they say:
>>>>>
>>>>> *Numbering*
>>>>> The standard e.164 numbering plan (ITU <http://www.itu.int/>) is 
>>>>> used. North American numbers are required to be prefixed with a 
>>>>> '1'. International numbers need to be prefixed with "011".
>>>>>
>>>>>
>>>>> I think they are confused about what is E164....
>>>>>
>>>>> You should try to prefix with 1 as Michael recommended.
>>>>>
>>>>> David Ponzone Direction Technique
>>>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>>>> tel:      01 74 03 18 97
>>>>> gsm:   06 66 98 76 34
>>>>>
>>>>> Service ClientIPeva
>>>>> tel:      0811 46 26 26
>>>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - 
>>>>> www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/>
>>>>>
>>>>> /Ce message et toutes les pi?ces jointes sont confidentiels et 
>>>>> ?tablis ? l'intention exclusive de ses destinataires. Toute 
>>>>> utilisation ou diffusion non autoris?e est interdite. Tout message 
>>>>> ?lectronique est susceptible d'alt?ration. /*/IPeva/*/ d?cline 
>>>>> toute responsabilit? au titre de ce message s'il a ?t? alt?r?, 
>>>>> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce 
>>>>> message, merci de le d?truire imm?diatement et d'avertir 
>>>>> l'exp?diteur./
>>>>> /
>>>>> /
>>>>>
>>>>>
>>>>>
>>>>> Le 23/06/2011 ? 08:19, Michael Collins a ?crit :
>>>>>
>>>>>> Perhaps the gateway wants the leading 1 or expects e.164 format?
>>>>>> -MC
>>>>>>
>>>>>> On Wed, Jun 22, 2011 at 5:37 PM, Brian West <brian at freeswitch.org 
>>>>>> <mailto:brian at freeswitch.org>> wrote:
>>>>>>
>>>>>>     484 is address incomplete means you didn't send enough digits
>>>>>>     to the gateway usually.
>>>>>>
>>>>>>     /b
>>>>>>
>>>>>>     On Jun 22, 2011, at 7:34 PM, Brad Mina wrote:
>>>>>>
>>>>>>>     Ah, thought it was recv not send. Good to know that an
>>>>>>>     invalid gateway issue
>>>>>>>     can cause this.
>>>>>>>
>>>>>>>     On Wed, Jun 22, 2011 at 5:31 PM, Michael Collins
>>>>>>>     <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
>>>>>>>
>>>>>>>>     The 484 I believe is a result of the bad gateway. In any
>>>>>>>>     case, the OP needs
>>>>>>>>     to figure out what's up w/ that gw or no calls will be
>>>>>>>>     going out...
>>>>>>>>
>>>>>>>>     -MC
>>>>>>>
>>>>>>
>>>>>>
>>>>>>     _______________________________________________
>>>>>>     Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>     http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>>>>
>>>>>>     FreeSWITCH-users mailing list
>>>>>>     FreeSWITCH-users at lists.freeswitch.org
>>>>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org 
>>>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com  877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org 
>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com  877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
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From bmoore at statirasystems.com  Thu Jun 23 21:15:35 2011
From: bmoore at statirasystems.com (William Moore)
Date: Thu, 23 Jun 2011 13:15:35 -0400
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <4E037099.9090107@statirasystems.com>
References: <4E0219B2.9070808@statirasystems.com>	<4E0227C8.9080304@statirasystems.com>	<4E02340D.2050708@statirasystems.com>	<BANLkTiksPsLhd23O-ws626svwnKcJJRmXw@mail.gmail.com>	<BANLkTinAjxG3Bp7jy3GTRs1km_DdoFDrCA@mail.gmail.com>	<BANLkTimip1BOfOG+U9rBOAJeiUJK_PL4fw@mail.gmail.com>	<BANLkTimBC4FKan=kjNoizbGUFGAe1ACu-A@mail.gmail.com>	<AC8EC5A4-1799-433A-9DD0-A715ADF5F256@freeswitch.org>	<BANLkTikQmkxi_C_1aQocH3O+nQrt4eaA9A@mail.gmail.com>	<B4D334F2-EE02-4EFB-A4A3-165C55CF7606@ipeva.fr>	<4E03669C.2030603@statirasystems.com>	<963C184F-7466-4874-80F1-0CC13FEC8581@ipeva.fr>	<4E036EE5.60208@statirasystems.com>	<4E036F3B.8090303@statirasystems.com>
	<4E037099.9090107@statirasystems.com>
Message-ID: <4E0374B7.8030709@statirasystems.com>

Ok, I reregistered and called right away. I get the following:

   0.000000 66.227.100.20 ->  192.168.1.254 SIP/SDP Request: INVITE sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with session description
   0.000766 192.168.1.254 ->  66.227.100.20 SIP Status: 100 Trying
   0.002321 192.168.1.254 ->  66.227.100.20 SIP Status: 407 Proxy Authentication Required
   0.021120 66.227.100.20 ->  192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
   3.272168 192.168.1.254 ->  66.227.100.20 SIP Request: REGISTER sip:jnctn.net;transport=udp
   3.292076 66.227.100.20 ->  192.168.1.254 SIP Status: 200 OK    (1 bindings)

The call just states it is a none working number. That means something in my dial plan is incorrect. I am double checking it now.

William J. Moore
Statira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com


On 06/23/2011 12:58 PM, William Moore wrote:
> When I put it in and it reloads, blue.box removes it. Can I reference 
> the gateway in another sip interface file or does it have to be in 
> that one?
> William J. Moore
> Statira Systems
> 611 Caroline St
> Fredericksburg, VA
> http://www.statirasystems.com
>
> On 06/23/2011 12:52 PM, William Moore wrote:
>> correction: missed a line.
>>
>>   0.000000 66.227.100.20 -> 192.168.1.254 SIP/SDP Request: INVITE 
>> sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with 
>> session description
>>   0.000814 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>>   0.002440 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy 
>> Authentication Required
>>   0.023173 66.227.100.20 -> 192.168.1.254 SIP Request: ACK 
>> sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>
>> William J. Moore
>> Statira Systems
>> 611 Caroline St
>> Fredericksburg, VA
>> http://www.statirasystems.com
>>
>> On 06/23/2011 12:50 PM, William Moore wrote:
>>> I get this with wireshark
>>>
>>>  40.345679 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>>>  40.347381 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy 
>>> Authentication Required
>>>  40.367906 66.227.100.20 -> 192.168.1.254 SIP Request: ACK 
>>> sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>>
>>> I am looking for the expiration now..
>>> William J. Moore
>>> Statira Systems
>>> 611 Caroline St
>>> Fredericksburg, VA
>>> http://www.statirasystems.com
>>>
>>> On 06/23/2011 12:25 PM, David Ponzone wrote:
>>>> Take a sip trace on your box with tcpdump or preferably, tshark:
>>>> tshark port 5080
>>>>
>>>> You will then see if you receive it.
>>>>
>>>> In the config you sent, I don't see anything about register 
>>>> expiration, or NAT keepalive, so I would really recommend you add a 
>>>> ping every 30 seconds to your gateway config.
>>>>
>>>> If the issue is there, it's quite easy to see:
>>>> unregister the gateway
>>>> register it again
>>>> make an inbound call right away
>>>>
>>>> David Ponzone Direction Technique
>>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>>> tel:      01 74 03 18 97
>>>> gsm:   06 66 98 76 34
>>>>
>>>> Service ClientIPeva
>>>> tel:      0811 46 26 26
>>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - 
>>>> www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/>
>>>>
>>>> /Ce message et toutes les pi?ces jointes sont confidentiels et 
>>>> ?tablis ? l'intention exclusive de ses destinataires. Toute 
>>>> utilisation ou diffusion non autoris?e est interdite. Tout message 
>>>> ?lectronique est susceptible d'alt?ration. /*/IPeva/*/ d?cline 
>>>> toute responsabilit? au titre de ce message s'il a ?t? alt?r?, 
>>>> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, 
>>>> merci de le d?truire imm?diatement et d'avertir l'exp?diteur./
>>>> /
>>>> /
>>>>
>>>>
>>>>
>>>> Le 23/06/2011 ? 18:15, William Moore a ?crit :
>>>>
>>>>> It was the 1 and it needs to be set for e.164(+1). Blue.box makes 
>>>>> it a little more difficult since it wants to manage the xmls. I 
>>>>> could have set it up as a separate config file however that 
>>>>> defeats the purpose of the GUI.
>>>>>
>>>>> I still have the issue of incoming calls not even showing up in 
>>>>> sofia. Junction Networks says it is going out, but my server is 
>>>>> not responding. I'm not sure if I have the settings for incoming 
>>>>> correct.
>>>>> William J. Moore
>>>>> Statira Systems
>>>>> 611 Caroline St
>>>>> Fredericksburg, VA
>>>>> http://www.statirasystems.com
>>>>>
>>>>> On 06/23/2011 04:21 AM, David Ponzone wrote:
>>>>>> William,
>>>>>>
>>>>>> If I believe the example configuraiton on Junction's web, they 
>>>>>> don't expect E164:
>>>>>>
>>>>>>     <extension name="junctionnetworks.com  <http://junctionnetworks.com/>>
>>>>>>      <condition field="destination_number" expression="^(1{0,1}\d{10})$">
>>>>>>        <action application="set"data="effective_caller_id_number=OUTBOUNDCALLERID"/>
>>>>>>        <action application="bridge" data="sofia/gateway/junctionnetworks.com/$1"/>
>>>>>>       </condition>
>>>>>>     </extension>
>>>>>> But, elsewhere in their knowledgebase, they say:
>>>>>>
>>>>>> *Numbering*
>>>>>> The standard e.164 numbering plan (ITU <http://www.itu.int/>) is 
>>>>>> used. North American numbers are required to be prefixed with a 
>>>>>> '1'. International numbers need to be prefixed with "011".
>>>>>>
>>>>>>
>>>>>> I think they are confused about what is E164....
>>>>>>
>>>>>> You should try to prefix with 1 as Michael recommended.
>>>>>>
>>>>>> David Ponzone Direction Technique
>>>>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>>>>> tel:      01 74 03 18 97
>>>>>> gsm:   06 66 98 76 34
>>>>>>
>>>>>> Service ClientIPeva
>>>>>> tel:      0811 46 26 26
>>>>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - 
>>>>>> www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/>
>>>>>>
>>>>>> /Ce message et toutes les pi?ces jointes sont confidentiels et 
>>>>>> ?tablis ? l'intention exclusive de ses destinataires. Toute 
>>>>>> utilisation ou diffusion non autoris?e est interdite. Tout 
>>>>>> message ?lectronique est susceptible d'alt?ration. 
>>>>>> /*/IPeva/*/ d?cline toute responsabilit? au titre de ce message 
>>>>>> s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas 
>>>>>> destinataire de ce message, merci de le d?truire imm?diatement et 
>>>>>> d'avertir l'exp?diteur./
>>>>>> /
>>>>>> /
>>>>>>
>>>>>>
>>>>>>
>>>>>> Le 23/06/2011 ? 08:19, Michael Collins a ?crit :
>>>>>>
>>>>>>> Perhaps the gateway wants the leading 1 or expects e.164 format?
>>>>>>> -MC
>>>>>>>
>>>>>>> On Wed, Jun 22, 2011 at 5:37 PM, Brian West 
>>>>>>> <brian at freeswitch.org <mailto:brian at freeswitch.org>> wrote:
>>>>>>>
>>>>>>>     484 is address incomplete means you didn't send enough
>>>>>>>     digits to the gateway usually.
>>>>>>>
>>>>>>>     /b
>>>>>>>
>>>>>>>     On Jun 22, 2011, at 7:34 PM, Brad Mina wrote:
>>>>>>>
>>>>>>>>     Ah, thought it was recv not send. Good to know that an
>>>>>>>>     invalid gateway issue
>>>>>>>>     can cause this.
>>>>>>>>
>>>>>>>>     On Wed, Jun 22, 2011 at 5:31 PM, Michael Collins
>>>>>>>>     <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
>>>>>>>>
>>>>>>>>>     The 484 I believe is a result of the bad gateway. In any
>>>>>>>>>     case, the OP needs
>>>>>>>>>     to figure out what's up w/ that gw or no calls will be
>>>>>>>>>     going out...
>>>>>>>>>
>>>>>>>>>     -MC
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>     _______________________________________________
>>>>>>>     Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>     http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>>>>>
>>>>>>>     FreeSWITCH-users mailing list
>>>>>>>     FreeSWITCH-users at lists.freeswitch.org
>>>>>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org 
>>>>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com  877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org 
>>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com  877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From anthony.minessale at gmail.com  Thu Jun 23 21:25:47 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 23 Jun 2011 12:25:47 -0500
Subject: [Freeswitch-users] mod_rtmp client API
In-Reply-To: <4E03603B.6020405@livecall.com>
References: <801B6B74CA73484F96B9BBD9BE125561@e1705>
	<BANLkTikmOfC7ot12-sGcO3aZ7+bH_cyGCQ@mail.gmail.com>
	<4E03603B.6020405@livecall.com>
Message-ID: <BANLkTikAxr_SOnAAE8rbPkdhLDmXhn7cLA@mail.gmail.com>

lets open a new feature JIRA on it and assemble a list of participants
and we'll discuss it at next Wednesday public meeting, this thread and
IRC.


On Thu, Jun 23, 2011 at 10:48 AM, Jack <jack at livecall.com> wrote:
> I am interested.... ?It looks like a clean, simple way to connect to
> Freeswitch.
> Nice Job!
> Jack
>
> On 6/22/2011 9:19 AM, Anthony Minessale wrote:
>> Actually......
>>
>> We have something we are working on that we are considering releasing
>> if we can build a community base effort around it.
>> When I get a sign of enough interest we'll go from there.
>>
>> It would be nice to do some presentations at ClueCon about it.
>>
>>
>> On Wed, Jun 22, 2011 at 10:53 AM, Madovsky<infos at madovsky.org> ?wrote:
>>> if I type "actionscript mod_rtmp freeswitch" on google
>>> nothing that helps a common developer to work on mod_rtmp
>>> ....
>>>
>>>
>>> ----- Original Message -----
>>> From: Madovsky
>>> To: FreeSWITCH Users Help
>>> Sent: Wednesday, June 22, 2011 11:20 AM
>>> Subject: Re: [Freeswitch-users] mod_rtmp client API
>>> Mike,
>>>
>>> I didn't ask tools, but to know at least which function it needs to call to
>>> the server
>>> to make it work ! :)
>>>
>>> ----- Original Message -----
>>> From: Michael Collins
>>> To: FreeSWITCH Users Help
>>> Sent: Wednesday, June 22, 2011 11:11 AM
>>> Subject: Re: [Freeswitch-users] mod_rtmp client API
>>> No, it's a DIY project. We provided the server, you provide the client. :D
>>> Google around and you will find various tools and pieces to start building
>>> your own.
>>> -MC
>>>
>>> On Wed, Jun 22, 2011 at 7:52 AM, Madovsky<infos at madovsky.org> ?wrote:
>>>> Is there any client side API to use mod_rtmp ?
>>>>
>>>> Thanks
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>> ________________________________
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From david.ponzone at ipeva.fr  Thu Jun 23 21:31:28 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 23 Jun 2011 19:31:28 +0200
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <4E0374B7.8030709@statirasystems.com>
References: <4E0219B2.9070808@statirasystems.com>	<4E0227C8.9080304@statirasystems.com>	<4E02340D.2050708@statirasystems.com>	<BANLkTiksPsLhd23O-ws626svwnKcJJRmXw@mail.gmail.com>	<BANLkTinAjxG3Bp7jy3GTRs1km_DdoFDrCA@mail.gmail.com>	<BANLkTimip1BOfOG+U9rBOAJeiUJK_PL4fw@mail.gmail.com>	<BANLkTimBC4FKan=kjNoizbGUFGAe1ACu-A@mail.gmail.com>	<AC8EC5A4-1799-433A-9DD0-A715ADF5F256@freeswitch.org>	<BANLkTikQmkxi_C_1aQocH3O+nQrt4eaA9A@mail.gmail.com>	<B4D334F2-EE02-4EFB-A4A3-165C55CF7606@ipeva.fr>	<4E03669C.2030603@statirasystems.com>	<963C184F-7466-4874-80F1-0CC13FEC8581@ipeva.fr>	<4E036EE5.60208@statirasystems.com>	<4E036F3B.8090303@statirasystems.com>
	<4E037099.9090107@statirasystems.com>
	<4E0374B7.8030709@statirasystems.com>
Message-ID: <F2D1C5FE-6BF3-4537-AB64-F47004DBA73F@ipeva.fr>

The issue is that you are expecting them to authenticate, but they won't.

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

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

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




Le 23/06/2011 ? 19:15, William Moore a ?crit :

> Ok, I reregistered and called right away. I get the following:
>   0.000000 66.227.100.20 -> 192.168.1.254 SIP/SDP Request: INVITE sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with session description
>   0.000766 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>   0.002321 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy Authentication Required
>   0.021120 66.227.100.20 -> 192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>   3.272168 192.168.1.254 -> 66.227.100.20 SIP Request: REGISTER sip:jnctn.net;transport=udp
>   3.292076 66.227.100.20 -> 192.168.1.254 SIP Status: 200 OK    (1 bindings)
> 
> The call just states it is a none working number. That means something in my dial plan is incorrect. I am double checking it now.
> William J. Moore
> Statira Systems
> 611 Caroline St
> Fredericksburg, VA
> http://www.statirasystems.com
> 
> On 06/23/2011 12:58 PM, William Moore wrote:
>> 
>> When I put it in and it reloads, blue.box removes it. Can I reference the gateway in another sip interface file or does it have to be in that one?
>> William J. Moore
>> Statira Systems
>> 611 Caroline St
>> Fredericksburg, VA
>> http://www.statirasystems.com
>> 
>> On 06/23/2011 12:52 PM, William Moore wrote:
>>> 
>>> correction: missed a line.
>>> 
>>>   0.000000 66.227.100.20 -> 192.168.1.254 SIP/SDP Request: INVITE sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with session description
>>>   0.000814 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>>>   0.002440 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy Authentication Required
>>>   0.023173 66.227.100.20 -> 192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>> 
>>> William J. Moore
>>> Statira Systems
>>> 611 Caroline St
>>> Fredericksburg, VA
>>> http://www.statirasystems.com
>>> 
>>> On 06/23/2011 12:50 PM, William Moore wrote:
>>>> 
>>>> I get this with wireshark
>>>> 
>>>>  40.345679 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>>>>  40.347381 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy Authentication Required
>>>>  40.367906 66.227.100.20 -> 192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>>> 
>>>> I am looking for the expiration now..
>>>> William J. Moore
>>>> Statira Systems
>>>> 611 Caroline St
>>>> Fredericksburg, VA
>>>> http://www.statirasystems.com
>>>> 
>>>> On 06/23/2011 12:25 PM, David Ponzone wrote:
>>>>> 
>>>>> Take a sip trace on your box with tcpdump or preferably, tshark:
>>>>> tshark port 5080
>>>>> 
>>>>> You will then see if you receive it.
>>>>> 
>>>>> In the config you sent, I don't see anything about register expiration, or NAT keepalive, so I would really recommend you add a ping every 30 seconds to your gateway config.
>>>>> 
>>>>> If the issue is there, it's quite easy to see:
>>>>> unregister the gateway
>>>>> register it again
>>>>> make an inbound call right away
>>>>> 
>>>>> David Ponzone  Direction Technique
>>>>> email: david.ponzone at ipeva.fr
>>>>> tel:      01 74 03 18 97
>>>>> gsm:   06 66 98 76 34
>>>>> 
>>>>> Service Client IPeva
>>>>> tel:      0811 46 26 26
>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>> 
>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Le 23/06/2011 ? 18:15, William Moore a ?crit :
>>>>> 
>>>>>> It was the 1 and it needs to be set for e.164(+1). Blue.box makes it a little more difficult since it wants to manage the xmls. I could have set it up as a separate config file however that defeats the purpose of the GUI.
>>>>>> 
>>>>>> I still have the issue of incoming calls not even showing up in sofia. Junction Networks says it is going out, but my server is not responding. I'm not sure if I have the settings for incoming correct.
>>>>>> William J. Moore
>>>>>> Statira Systems
>>>>>> 611 Caroline St
>>>>>> Fredericksburg, VA
>>>>>> http://www.statirasystems.com
>>>>>> 
>>>>>> On 06/23/2011 04:21 AM, David Ponzone wrote:
>>>>>>> 
>>>>>>> William,
>>>>>>> 
>>>>>>> If I believe the example configuraiton on Junction's web, they don't expect E164:
>>>>>>> 
>>>>>>>     <extension name="junctionnetworks.com>
>>>>>>>     <condition field="destination_number" expression="^(1{0,1}\d{10})$">
>>>>>>>       <action application="set"data="effective_caller_id_number=OUTBOUNDCALLERID"/>
>>>>>>>       <action application="bridge" data="sofia/gateway/junctionnetworks.com/$1"/>
>>>>>>>      </condition>
>>>>>>>    </extension>
>>>>>>> But, elsewhere in their knowledgebase, they say:
>>>>>>> 
>>>>>>> Numbering
>>>>>>> The standard e.164 numbering plan (ITU) is used. North American numbers are required to be prefixed with a '1'. International numbers need to be prefixed with "011".
>>>>>>> 
>>>>>>> 
>>>>>>> I think they are confused about what is E164....
>>>>>>> 
>>>>>>> You should try to prefix with 1 as Michael recommended.
>>>>>>> 
>>>>>>> David Ponzone  Direction Technique
>>>>>>> email: david.ponzone at ipeva.fr
>>>>>>> tel:      01 74 03 18 97
>>>>>>> gsm:   06 66 98 76 34
>>>>>>> 
>>>>>>> Service Client IPeva
>>>>>>> tel:      0811 46 26 26
>>>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>>>> 
>>>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> Le 23/06/2011 ? 08:19, Michael Collins a ?crit :
>>>>>>> 
>>>>>>>> Perhaps the gateway wants the leading 1 or expects e.164 format?
>>>>>>>> -MC
>>>>>>>> 
>>>>>>>> On Wed, Jun 22, 2011 at 5:37 PM, Brian West <brian at freeswitch.org> wrote:
>>>>>>>> 484 is address incomplete means you didn't send enough digits to the gateway usually.
>>>>>>>> 
>>>>>>>> /b
>>>>>>>> 
>>>>>>>> On Jun 22, 2011, at 7:34 PM, Brad Mina wrote:
>>>>>>>> 
>>>>>>>>> Ah, thought it was recv not send. Good to know that an invalid gateway issue
>>>>>>>>> can cause this.
>>>>>>>>> 
>>>>>>>>> On Wed, Jun 22, 2011 at 5:31 PM, Michael Collins <msc at freeswitch.org> wrote:
>>>>>>>>> 
>>>>>>>>>> The 484 I believe is a result of the bad gateway. In any case, the OP needs
>>>>>>>>>> to figure out what's up w/ that gw or no calls will be going out...
>>>>>>>>>> 
>>>>>>>>>> -MC
>>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>> 
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>> 
>>>>>>>> 
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>> 
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at ipeva.fr  Thu Jun 23 21:34:22 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 23 Jun 2011 19:34:22 +0200
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <F2D1C5FE-6BF3-4537-AB64-F47004DBA73F@ipeva.fr>
References: <4E0219B2.9070808@statirasystems.com>	<4E0227C8.9080304@statirasystems.com>	<4E02340D.2050708@statirasystems.com>	<BANLkTiksPsLhd23O-ws626svwnKcJJRmXw@mail.gmail.com>	<BANLkTinAjxG3Bp7jy3GTRs1km_DdoFDrCA@mail.gmail.com>	<BANLkTimip1BOfOG+U9rBOAJeiUJK_PL4fw@mail.gmail.com>	<BANLkTimBC4FKan=kjNoizbGUFGAe1ACu-A@mail.gmail.com>	<AC8EC5A4-1799-433A-9DD0-A715ADF5F256@freeswitch.org>	<BANLkTikQmkxi_C_1aQocH3O+nQrt4eaA9A@mail.gmail.com>	<B4D334F2-EE02-4EFB-A4A3-165C55CF7606@ipeva.fr>	<4E03669C.2030603@statirasystems.com>	<963C184F-7466-4874-80F1-0CC13FEC8581@ipeva.fr>	<4E036EE5.60208@statirasystems.com>	<4E036F3B.8090303@statirasystems.com>
	<4E037099.9090107@statirasystems.com>
	<4E0374B7.8030709@statirasystems.com>
	<F2D1C5FE-6BF3-4537-AB64-F47004DBA73F@ipeva.fr>
Message-ID: <F7B1B3E9-F6E4-4CF5-8A44-994E4EE1B902@ipeva.fr>

In sipinterface_2, make the following modification:

 <param name="auth-calls" value="false"/>	

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

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

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




Le 23/06/2011 ? 19:31, David Ponzone a ?crit :

> The issue is that you are expecting them to authenticate, but they won't.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 23/06/2011 ? 19:15, William Moore a ?crit :
> 
>> Ok, I reregistered and called right away. I get the following:
>>   0.000000 66.227.100.20 -> 192.168.1.254 SIP/SDP Request: INVITE sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with session description
>>   0.000766 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>>   0.002321 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy Authentication Required
>>   0.021120 66.227.100.20 -> 192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>   3.272168 192.168.1.254 -> 66.227.100.20 SIP Request: REGISTER sip:jnctn.net;transport=udp
>>   3.292076 66.227.100.20 -> 192.168.1.254 SIP Status: 200 OK    (1 bindings)
>> 
>> The call just states it is a none working number. That means something in my dial plan is incorrect. I am double checking it now.
>> William J. Moore
>> Statira Systems
>> 611 Caroline St
>> Fredericksburg, VA
>> http://www.statirasystems.com
>> 
>> On 06/23/2011 12:58 PM, William Moore wrote:
>>> 
>>> When I put it in and it reloads, blue.box removes it. Can I reference the gateway in another sip interface file or does it have to be in that one?
>>> William J. Moore
>>> Statira Systems
>>> 611 Caroline St
>>> Fredericksburg, VA
>>> http://www.statirasystems.com
>>> 
>>> On 06/23/2011 12:52 PM, William Moore wrote:
>>>> 
>>>> correction: missed a line.
>>>> 
>>>>   0.000000 66.227.100.20 -> 192.168.1.254 SIP/SDP Request: INVITE sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with session description
>>>>   0.000814 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>>>>   0.002440 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy Authentication Required
>>>>   0.023173 66.227.100.20 -> 192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>>> 
>>>> William J. Moore
>>>> Statira Systems
>>>> 611 Caroline St
>>>> Fredericksburg, VA
>>>> http://www.statirasystems.com
>>>> 
>>>> On 06/23/2011 12:50 PM, William Moore wrote:
>>>>> 
>>>>> I get this with wireshark
>>>>> 
>>>>>  40.345679 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>>>>>  40.347381 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy Authentication Required
>>>>>  40.367906 66.227.100.20 -> 192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>>>> 
>>>>> I am looking for the expiration now..
>>>>> William J. Moore
>>>>> Statira Systems
>>>>> 611 Caroline St
>>>>> Fredericksburg, VA
>>>>> http://www.statirasystems.com
>>>>> 
>>>>> On 06/23/2011 12:25 PM, David Ponzone wrote:
>>>>>> 
>>>>>> Take a sip trace on your box with tcpdump or preferably, tshark:
>>>>>> tshark port 5080
>>>>>> 
>>>>>> You will then see if you receive it.
>>>>>> 
>>>>>> In the config you sent, I don't see anything about register expiration, or NAT keepalive, so I would really recommend you add a ping every 30 seconds to your gateway config.
>>>>>> 
>>>>>> If the issue is there, it's quite easy to see:
>>>>>> unregister the gateway
>>>>>> register it again
>>>>>> make an inbound call right away
>>>>>> 
>>>>>> David Ponzone  Direction Technique
>>>>>> email: david.ponzone at ipeva.fr
>>>>>> tel:      01 74 03 18 97
>>>>>> gsm:   06 66 98 76 34
>>>>>> 
>>>>>> Service Client IPeva
>>>>>> tel:      0811 46 26 26
>>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>>> 
>>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Le 23/06/2011 ? 18:15, William Moore a ?crit :
>>>>>> 
>>>>>>> It was the 1 and it needs to be set for e.164(+1). Blue.box makes it a little more difficult since it wants to manage the xmls. I could have set it up as a separate config file however that defeats the purpose of the GUI.
>>>>>>> 
>>>>>>> I still have the issue of incoming calls not even showing up in sofia. Junction Networks says it is going out, but my server is not responding. I'm not sure if I have the settings for incoming correct.
>>>>>>> William J. Moore
>>>>>>> Statira Systems
>>>>>>> 611 Caroline St
>>>>>>> Fredericksburg, VA
>>>>>>> http://www.statirasystems.com
>>>>>>> 
>>>>>>> On 06/23/2011 04:21 AM, David Ponzone wrote:
>>>>>>>> 
>>>>>>>> William,
>>>>>>>> 
>>>>>>>> If I believe the example configuraiton on Junction's web, they don't expect E164:
>>>>>>>> 
>>>>>>>>     <extension name="junctionnetworks.com>
>>>>>>>>     <condition field="destination_number" expression="^(1{0,1}\d{10})$">
>>>>>>>>       <action application="set"data="effective_caller_id_number=OUTBOUNDCALLERID"/>
>>>>>>>>       <action application="bridge" data="sofia/gateway/junctionnetworks.com/$1"/>
>>>>>>>>      </condition>
>>>>>>>>    </extension>
>>>>>>>> But, elsewhere in their knowledgebase, they say:
>>>>>>>> 
>>>>>>>> Numbering
>>>>>>>> The standard e.164 numbering plan (ITU) is used. North American numbers are required to be prefixed with a '1'. International numbers need to be prefixed with "011".
>>>>>>>> 
>>>>>>>> 
>>>>>>>> I think they are confused about what is E164....
>>>>>>>> 
>>>>>>>> You should try to prefix with 1 as Michael recommended.
>>>>>>>> 
>>>>>>>> David Ponzone  Direction Technique
>>>>>>>> email: david.ponzone at ipeva.fr
>>>>>>>> tel:      01 74 03 18 97
>>>>>>>> gsm:   06 66 98 76 34
>>>>>>>> 
>>>>>>>> Service Client IPeva
>>>>>>>> tel:      0811 46 26 26
>>>>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>>>>> 
>>>>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> Le 23/06/2011 ? 08:19, Michael Collins a ?crit :
>>>>>>>> 
>>>>>>>>> Perhaps the gateway wants the leading 1 or expects e.164 format?
>>>>>>>>> -MC
>>>>>>>>> 
>>>>>>>>> On Wed, Jun 22, 2011 at 5:37 PM, Brian West <brian at freeswitch.org> wrote:
>>>>>>>>> 484 is address incomplete means you didn't send enough digits to the gateway usually.
>>>>>>>>> 
>>>>>>>>> /b
>>>>>>>>> 
>>>>>>>>> On Jun 22, 2011, at 7:34 PM, Brad Mina wrote:
>>>>>>>>> 
>>>>>>>>>> Ah, thought it was recv not send. Good to know that an invalid gateway issue
>>>>>>>>>> can cause this.
>>>>>>>>>> 
>>>>>>>>>> On Wed, Jun 22, 2011 at 5:31 PM, Michael Collins <msc at freeswitch.org> wrote:
>>>>>>>>>> 
>>>>>>>>>>> The 484 I believe is a result of the bad gateway. In any case, the OP needs
>>>>>>>>>>> to figure out what's up w/ that gw or no calls will be going out...
>>>>>>>>>>> 
>>>>>>>>>>> -MC
>>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>> 
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>> 
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>> 
>>>>>>>> 
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>> 
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>> 
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 

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From philippe.sultan at gmail.com  Thu Jun 23 21:41:13 2011
From: philippe.sultan at gmail.com (Philippe Sultan)
Date: Thu, 23 Jun 2011 19:41:13 +0200
Subject: [Freeswitch-users] mod_rtmp client API
In-Reply-To: <BANLkTikAxr_SOnAAE8rbPkdhLDmXhn7cLA@mail.gmail.com>
References: <801B6B74CA73484F96B9BBD9BE125561@e1705>
	<BANLkTikmOfC7ot12-sGcO3aZ7+bH_cyGCQ@mail.gmail.com>
	<4E03603B.6020405@livecall.com>
	<BANLkTikAxr_SOnAAE8rbPkdhLDmXhn7cLA@mail.gmail.com>
Message-ID: <BANLkTimG0gN8zQzBA=pkQqDq4Q5FdYCguw@mail.gmail.com>

I'm very interested too. We have a Red5 server here and some AS3 audio
apps we'd love to test with mod_rtmp.

Thanks,

Philippe

On Thu, Jun 23, 2011 at 7:25 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> lets open a new feature JIRA on it and assemble a list of participants
> and we'll discuss it at next Wednesday public meeting, this thread and
> IRC.
>
>
> On Thu, Jun 23, 2011 at 10:48 AM, Jack <jack at livecall.com> wrote:
>> I am interested.... ?It looks like a clean, simple way to connect to
>> Freeswitch.
>> Nice Job!
>> Jack
>>
>> On 6/22/2011 9:19 AM, Anthony Minessale wrote:
>>> Actually......
>>>
>>> We have something we are working on that we are considering releasing
>>> if we can build a community base effort around it.
>>> When I get a sign of enough interest we'll go from there.
>>>
>>> It would be nice to do some presentations at ClueCon about it.
>>>
>>>
>>> On Wed, Jun 22, 2011 at 10:53 AM, Madovsky<infos at madovsky.org> ?wrote:
>>>> if I type "actionscript mod_rtmp freeswitch" on google
>>>> nothing that helps a common developer to work on mod_rtmp
>>>> ....
>>>>
>>>>
>>>> ----- Original Message -----
>>>> From: Madovsky
>>>> To: FreeSWITCH Users Help
>>>> Sent: Wednesday, June 22, 2011 11:20 AM
>>>> Subject: Re: [Freeswitch-users] mod_rtmp client API
>>>> Mike,
>>>>
>>>> I didn't ask tools, but to know at least which function it needs to call to
>>>> the server
>>>> to make it work ! :)
>>>>
>>>> ----- Original Message -----
>>>> From: Michael Collins
>>>> To: FreeSWITCH Users Help
>>>> Sent: Wednesday, June 22, 2011 11:11 AM
>>>> Subject: Re: [Freeswitch-users] mod_rtmp client API
>>>> No, it's a DIY project. We provided the server, you provide the client. :D
>>>> Google around and you will find various tools and pieces to start building
>>>> your own.
>>>> -MC
>>>>
>>>> On Wed, Jun 22, 2011 at 7:52 AM, Madovsky<infos at madovsky.org> ?wrote:
>>>>> Is there any client side API to use mod_rtmp ?
>>>>>
>>>>> Thanks
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>> ________________________________
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Philippe Sultan


From msc at freeswitch.org  Thu Jun 23 21:45:25 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 23 Jun 2011 10:45:25 -0700
Subject: [Freeswitch-users] mod_rtmp client API
In-Reply-To: <BANLkTikAxr_SOnAAE8rbPkdhLDmXhn7cLA@mail.gmail.com>
References: <801B6B74CA73484F96B9BBD9BE125561@e1705>
	<BANLkTikmOfC7ot12-sGcO3aZ7+bH_cyGCQ@mail.gmail.com>
	<4E03603B.6020405@livecall.com>
	<BANLkTikAxr_SOnAAE8rbPkdhLDmXhn7cLA@mail.gmail.com>
Message-ID: <BANLkTinMpTenRibLCt3u-FEXn3Cj0F+Zkw@mail.gmail.com>

On Thu, Jun 23, 2011 at 10:25 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> lets open a new feature JIRA on it and assemble a list of participants
> and we'll discuss it at next Wednesday public meeting, this thread and
> IRC.
>

http://jira.freeswitch.org/browse/FS-3368

All those who have a specific interest in working on the mod_rtmp
client-side features please go to that Jira and add your comments,
experience, etc. If you don't have any specific development experience with
Flash/RTMP but have the ability to do QC and testing then please add your
name to the hat - we can never have too many people doing tests.

Thanks for supporting the project!

-MC
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From bmoore at statirasystems.com  Thu Jun 23 21:55:15 2011
From: bmoore at statirasystems.com (William Moore)
Date: Thu, 23 Jun 2011 13:55:15 -0400
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <F7B1B3E9-F6E4-4CF5-8A44-994E4EE1B902@ipeva.fr>
References: <4E0219B2.9070808@statirasystems.com>	<4E0227C8.9080304@statirasystems.com>	<4E02340D.2050708@statirasystems.com>	<BANLkTiksPsLhd23O-ws626svwnKcJJRmXw@mail.gmail.com>	<BANLkTinAjxG3Bp7jy3GTRs1km_DdoFDrCA@mail.gmail.com>	<BANLkTimip1BOfOG+U9rBOAJeiUJK_PL4fw@mail.gmail.com>	<BANLkTimBC4FKan=kjNoizbGUFGAe1ACu-A@mail.gmail.com>	<AC8EC5A4-1799-433A-9DD0-A715ADF5F256@freeswitch.org>	<BANLkTikQmkxi_C_1aQocH3O+nQrt4eaA9A@mail.gmail.com>	<B4D334F2-EE02-4EFB-A4A3-165C55CF7606@ipeva.fr>	<4E03669C.2030603@statirasystems.com>	<963C184F-7466-4874-80F1-0CC13FEC8581@ipeva.fr>	<4E036EE5.60208@statirasystems.com>	<4E036F3B.8090303@statirasystems.com>	<4E037099.9090107@statirasystems.com>	<4E0374B7.8030709@statirasystems.com>	<F2D1C5FE-6BF3-4537-AB64-F47004DBA73F@ipeva.fr>
	<F7B1B3E9-F6E4-4CF5-8A44-994E4EE1B902@ipeva.fr>
Message-ID: <4E037E03.2070408@statirasystems.com>

I give up on blue.box. They seem preoccupied by there other projects. 
Documentation does not explain enough. It is like describing a human by 
their parts only and not how it interacts. Peace out.

William J. Moore
Statira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com


On 06/23/2011 01:34 PM, David Ponzone wrote:
> In sipinterface_2, make the following modification:
>
> <param name="auth-calls" value="false"/>
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service ClientIPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
> diffusion non autoris?e est interdite. Tout message ?lectronique est 
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au 
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
> n'?tes pas destinataire de ce message, merci de le d?truire 
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 23/06/2011 ? 19:31, David Ponzone a ?crit :
>
>> The issue is that you are expecting them to authenticate, but they 
>> won't.
>>
>> David Ponzone Direction Technique
>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service ClientIPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
>> <BLOCKED::http://www.ipeva-studio.com/>
>>
>> /Ce message et toutes les pi?ces jointes sont confidentiels et 
>> ?tablis ? l'intention exclusive de ses destinataires. Toute 
>> utilisation ou diffusion non autoris?e est interdite. Tout message 
>> ?lectronique est susceptible d'alt?ration. /*/IPeva/*/ d?cline toute 
>> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou 
>> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le 
>> d?truire imm?diatement et d'avertir l'exp?diteur./
>> /
>> /
>>
>>
>>
>> Le 23/06/2011 ? 19:15, William Moore a ?crit :
>>
>>> Ok, I reregistered and called right away. I get the following:
>>>    0.000000 66.227.100.20 ->  192.168.1.254 SIP/SDP Request: INVITEsip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with session description
>>>    0.000766 192.168.1.254 ->  66.227.100.20 SIP Status: 100 Trying
>>>    0.002321 192.168.1.254 ->  66.227.100.20 SIP Status: 407 Proxy Authentication Required
>>>    0.021120 66.227.100.20 ->  192.168.1.254 SIP Request: ACKsip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>>    3.272168 192.168.1.254 ->  66.227.100.20 SIP Request: REGISTERsip:jnctn.net;transport=udp
>>>    3.292076 66.227.100.20 ->  192.168.1.254 SIP Status: 200 OK    (1 bindings)
>>>
>>> The call just states it is a none working number. That means something in my dial plan is incorrect. I am double checking it now.
>>> William J. Moore
>>> Statira Systems
>>> 611 Caroline St
>>> Fredericksburg, VA
>>> http://www.statirasystems.com
>>>
>>> On 06/23/2011 12:58 PM, William Moore wrote:
>>>> When I put it in and it reloads, blue.box removes it. Can I 
>>>> reference the gateway in another sip interface file or does it have 
>>>> to be in that one?
>>>> William J. Moore
>>>> Statira Systems
>>>> 611 Caroline St
>>>> Fredericksburg, VA
>>>> http://www.statirasystems.com
>>>>
>>>> On 06/23/2011 12:52 PM, William Moore wrote:
>>>>> correction: missed a line.
>>>>>
>>>>>   0.000000 66.227.100.20 -> 192.168.1.254 SIP/SDP Request: INVITE 
>>>>> sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with 
>>>>> session description
>>>>>   0.000814 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>>>>>   0.002440 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy 
>>>>> Authentication Required
>>>>>   0.023173 66.227.100.20 -> 192.168.1.254 SIP Request: ACK 
>>>>> sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>>>>
>>>>> William J. Moore
>>>>> Statira Systems
>>>>> 611 Caroline St
>>>>> Fredericksburg, VA
>>>>> http://www.statirasystems.com
>>>>>
>>>>> On 06/23/2011 12:50 PM, William Moore wrote:
>>>>>> I get this with wireshark
>>>>>>
>>>>>>  40.345679 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>>>>>>  40.347381 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy 
>>>>>> Authentication Required
>>>>>>  40.367906 66.227.100.20 -> 192.168.1.254 SIP Request: ACK 
>>>>>> sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>>>>>
>>>>>> I am looking for the expiration now..
>>>>>> William J. Moore
>>>>>> Statira Systems
>>>>>> 611 Caroline St
>>>>>> Fredericksburg, VA
>>>>>> http://www.statirasystems.com
>>>>>>
>>>>>> On 06/23/2011 12:25 PM, David Ponzone wrote:
>>>>>>> Take a sip trace on your box with tcpdump or preferably, tshark:
>>>>>>> tshark port 5080
>>>>>>>
>>>>>>> You will then see if you receive it.
>>>>>>>
>>>>>>> In the config you sent, I don't see anything about register 
>>>>>>> expiration, or NAT keepalive, so I would really recommend you 
>>>>>>> add a ping every 30 seconds to your gateway config.
>>>>>>>
>>>>>>> If the issue is there, it's quite easy to see:
>>>>>>> unregister the gateway
>>>>>>> register it again
>>>>>>> make an inbound call right away
>>>>>>>
>>>>>>> David Ponzone Direction Technique
>>>>>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>>>>>> tel:      01 74 03 18 97
>>>>>>> gsm:   06 66 98 76 34
>>>>>>>
>>>>>>> Service ClientIPeva
>>>>>>> tel:      0811 46 26 26
>>>>>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - 
>>>>>>> www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/>
>>>>>>>
>>>>>>> /Ce message et toutes les pi?ces jointes sont confidentiels et 
>>>>>>> ?tablis ? l'intention exclusive de ses destinataires. Toute 
>>>>>>> utilisation ou diffusion non autoris?e est interdite. Tout 
>>>>>>> message ?lectronique est susceptible d'alt?ration. 
>>>>>>> /*/IPeva/*/ d?cline toute responsabilit? au titre de ce message 
>>>>>>> s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas 
>>>>>>> destinataire de ce message, merci de le d?truire imm?diatement 
>>>>>>> et d'avertir l'exp?diteur./
>>>>>>> /
>>>>>>> /
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Le 23/06/2011 ? 18:15, William Moore a ?crit :
>>>>>>>
>>>>>>>> It was the 1 and it needs to be set for e.164(+1). Blue.box 
>>>>>>>> makes it a little more difficult since it wants to manage the 
>>>>>>>> xmls. I could have set it up as a separate config file however 
>>>>>>>> that defeats the purpose of the GUI.
>>>>>>>>
>>>>>>>> I still have the issue of incoming calls not even showing up in 
>>>>>>>> sofia. Junction Networks says it is going out, but my server is 
>>>>>>>> not responding. I'm not sure if I have the settings for 
>>>>>>>> incoming correct.
>>>>>>>> William J. Moore
>>>>>>>> Statira Systems
>>>>>>>> 611 Caroline St
>>>>>>>> Fredericksburg, VA
>>>>>>>> http://www.statirasystems.com
>>>>>>>>
>>>>>>>> On 06/23/2011 04:21 AM, David Ponzone wrote:
>>>>>>>>> William,
>>>>>>>>>
>>>>>>>>> If I believe the example configuraiton on Junction's web, they 
>>>>>>>>> don't expect E164:
>>>>>>>>>
>>>>>>>>>     <extension name="junctionnetworks.com  <http://junctionnetworks.com/>>
>>>>>>>>>      <condition field="destination_number" expression="^(1{0,1}\d{10})$">
>>>>>>>>>        <action application="set"data="effective_caller_id_number=OUTBOUNDCALLERID"/>
>>>>>>>>>        <action application="bridge" data="sofia/gateway/junctionnetworks.com/$1"/>
>>>>>>>>>       </condition>
>>>>>>>>>     </extension>
>>>>>>>>> But, elsewhere in their knowledgebase, they say:
>>>>>>>>>
>>>>>>>>> *Numbering*
>>>>>>>>> The standard e.164 numbering plan (ITU <http://www.itu.int/>) 
>>>>>>>>> is used. North American numbers are required to be prefixed 
>>>>>>>>> with a '1'. International numbers need to be prefixed with "011".
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> I think they are confused about what is E164....
>>>>>>>>>
>>>>>>>>> You should try to prefix with 1 as Michael recommended.
>>>>>>>>>
>>>>>>>>> David Ponzone Direction Technique
>>>>>>>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>>>>>>>> tel:      01 74 03 18 97
>>>>>>>>> gsm:   06 66 98 76 34
>>>>>>>>>
>>>>>>>>> Service ClientIPeva
>>>>>>>>> tel:      0811 46 26 26
>>>>>>>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - 
>>>>>>>>> www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/>
>>>>>>>>>
>>>>>>>>> /Ce message et toutes les pi?ces jointes sont confidentiels et 
>>>>>>>>> ?tablis ? l'intention exclusive de ses destinataires. Toute 
>>>>>>>>> utilisation ou diffusion non autoris?e est interdite. Tout 
>>>>>>>>> message ?lectronique est susceptible d'alt?ration. 
>>>>>>>>> /*/IPeva/*/ d?cline toute responsabilit? au titre de ce 
>>>>>>>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes 
>>>>>>>>> pas destinataire de ce message, merci de le d?truire 
>>>>>>>>> imm?diatement et d'avertir l'exp?diteur./
>>>>>>>>> /
>>>>>>>>> /
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Le 23/06/2011 ? 08:19, Michael Collins a ?crit :
>>>>>>>>>
>>>>>>>>>> Perhaps the gateway wants the leading 1 or expects e.164 format?
>>>>>>>>>> -MC
>>>>>>>>>>
>>>>>>>>>> On Wed, Jun 22, 2011 at 5:37 PM, Brian West 
>>>>>>>>>> <brian at freeswitch.org <mailto:brian at freeswitch.org>> wrote:
>>>>>>>>>>
>>>>>>>>>>     484 is address incomplete means you didn't send enough
>>>>>>>>>>     digits to the gateway usually.
>>>>>>>>>>
>>>>>>>>>>     /b
>>>>>>>>>>
>>>>>>>>>>     On Jun 22, 2011, at 7:34 PM, Brad Mina wrote:
>>>>>>>>>>
>>>>>>>>>>>     Ah, thought it was recv not send. Good to know that an
>>>>>>>>>>>     invalid gateway issue
>>>>>>>>>>>     can cause this.
>>>>>>>>>>>
>>>>>>>>>>>     On Wed, Jun 22, 2011 at 5:31 PM, Michael Collins
>>>>>>>>>>>     <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
>>>>>>>>>>>
>>>>>>>>>>>>     The 484 I believe is a result of the bad gateway. In
>>>>>>>>>>>>     any case, the OP needs
>>>>>>>>>>>>     to figure out what's up w/ that gw or no calls will be
>>>>>>>>>>>>     going out...
>>>>>>>>>>>>
>>>>>>>>>>>>     -MC
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>     _______________________________________________
>>>>>>>>>>     Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>     http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>>>>>>>>
>>>>>>>>>>     FreeSWITCH-users mailing list
>>>>>>>>>>     FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>> http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>>>>>>>>
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org 
>>>>>>>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> http://www.cluecon.com  877-7-4ACLUE
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org 
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>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com  877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
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>
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From msc at freeswitch.org  Thu Jun 23 22:32:56 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 23 Jun 2011 11:32:56 -0700
Subject: [Freeswitch-users] Call drops after 30 seconds
In-Reply-To: <28830F22-9654-45D2-B00A-1E69E32A42F9@mralston.com>
References: <71F1700A-7641-43E5-A7C6-F25A0DC2EBC5@mralston.com>
	<BANLkTikoP+hvtbzDn6fm_O6MmSN--x7TyA@mail.gmail.com>
	<C3BE3091-1685-42D1-ACD1-223B370607B7@mralston.com>
	<BANLkTi=4e1t607G0xCRygHGotxQrsXcnzw@mail.gmail.com>
	<28830F22-9654-45D2-B00A-1E69E32A42F9@mralston.com>
Message-ID: <BANLkTimBBN8RpAcEiJM6Rj6tegZSeydmUg@mail.gmail.com>

Matthew,

Did you get resolution on this one? I think you can go to the sip profile
and work with these lines:
    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="ext-sip-ip" value="auto-nat"/>

You can put an IP address in those and try it out. Let us know what happens.
-MC

On Wed, Jun 22, 2011 at 8:26 AM, Matthew Ralston <freeswitch at mralston.com>wrote:

> Hi Chris,
>
> The FreeSWITCH box does indeed have a private IP address. The Cisco Linksys
> WAG160N router in front of it has UPnP enabled, not that I particularly
> trust it.
>
> So maybe it's worth hard coding the public IP into the Sofia config.
>
> Where is the best place to put the public IP? The internal.xml and
> external.xml files both refer to $${local_ip_v4} although I don't see where
> this comes from. Does FreeSWITCH figure this out itself in some way?
>
> As I have internal SIP phones, an external SIP provider and (whilst
> debugging) some external SIP phones (which use the internal profile!!). I'm
> concerned that if I hard code the public IP address in to the wrong place it
> will cause problems for the internal SIP phones.
>
>
> Kind regards,
>
> Matthew Ralston
> Web Developer & IT Consultant
>
> matt at mralston.co.uk
> www.mralston.com
>
> On 22 Jun 2011, at 16:11, Chris Chen wrote:
>
> Hi Matthew, continue my last reply here
>
> 2) Is your FS server using private IP address?  you have to setup your FS
> external SIP/RTP IP address to the proper public IP address, by either using
> UPNP enabled router, STUN, or hardcoded public IP address in sofia profiles.
>
> Please check that.
>
> Thanks,
> Chris
>
>
> On Wed, Jun 22, 2011 at 10:49 AM, Matthew Ralston <freeswitch at mralston.com
> > wrote:
>
>> Hi Chris,
>>
>> Thanks for the quick reply!
>>
>> The FreeSWITCH box is in the DMZ of a Cisco Linksys WAG160N. Best I can
>> tell, the DMZ is doing its job and allowing all ports through, inbound and
>> outbound, TCP & UDP.
>>
>> The external SIP phones are a Cisco SPA504G and Bria on the iPhone. These
>> are behind behind a Cisco ASA5505, which has policy inspection for SIP
>> switched on, i.e. ALG. I have also tested with Bria going over 3G (so it's
>> not behind the Cisco ASA) and had the same problem.
>>
>> The other scenario we have is some Yealink T20P SIP phones on the same LAN
>> segment as the FreeSWITCH box. These can make A-leg only calls into
>> FreeSWITCH (like calling voicemail) and also calls to other internal SIP
>> phones fine. However when they make an outbound call the problem happens
>> again. In this case the b-leg of the calls are sent to an external SIP
>> provider and get cut after 30 seconds. Incidentally, we also use the same
>> SIP provider from an Asterisk box in our data centre and that doesn't have a
>> problem, so I believe the SIP provider is fine.
>>
>> Kind regards,
>>
>>
>> Matthew Ralston
>> Web Developer & IT Consultant
>>
>> matt at mralston.co.uk
>> www.mralston.com
>>
>> On 22 Jun 2011, at 14:21, Chris Chen wrote:
>>
>> Hi Matthew, this is typical behavior for the setup of SIP behind NAT.
>>
>>     1) Please provide the exact setup of remote SIP phones, what's the
>> router model, does it have SIP ALG enabled, what kind  of SIP phones
>>     2)
>>
>>
>> On Wed, Jun 22, 2011 at 8:38 AM, Matthew Ralston <freeswitch at mralston.com
>> > wrote:
>>
>>> Hi,
>>>
>>> I'm having a problem at the moment with calls being successfully set up,
>>> with two-way audio, being terminated by FreeSWITCH after 30 seconds.
>>>
>>> Internal calls (i.e. between SIP phones on the same LAN segment as the
>>> FreeSWITCH box) work flawlessly.
>>>
>>> The problem arises when at least one of the handsets is located elsewhere
>>> on the Internet. This behaviour is exhibited under the following
>>> circumstances:
>>>
>>> - A-leg only call, e.g. to voicemail when the handset is at another
>>> location on the Internet
>>> - A-leg-B-leg call if one or both of the handsets are at another location
>>> on the Internet
>>> - Inbound calls from our external SIP provider
>>> - Outbound calls to our external SIP provider
>>>
>>> So it is obvious that the problem is related to the SIP going via the
>>> Internet, but I'm having trouble understanding why.
>>>
>>> Whilst debugging this problem I have placed the FreeSWITCH box is in the
>>> DMZ on our router, so there should not be any ports blocked. The FreeSWITCH
>>> box itself is not running a software firewall.
>>>
>>> The calls themselves are absolutely fine for the first 30 seconds - each
>>> party can hear the other talking fine.
>>>
>>> The fact that the call is consistently dropped after 30 seconds (give or
>>> take a second or two for PDD) suggests that some timeout is being triggered.
>>>
>>> When FreeSWITCH terminates the call, the following is logged to the
>>> console:
>>>
>>> 2011-06-22 13:33:50.514941 [DEBUG] sofia.c:4787 Channel
>>> sofia/internal/1006 at public.ip.removed entering state [terminating][0]
>>> 2011-06-22 13:33:50.514941 [DEBUG] switch_channel.c:2641 (
>>> sofia/internal/1006 at public.ip.removed) Callstate Change ACTIVE -> HANGUP
>>> 2011-06-22 13:33:50.514941 [NOTICE] sofia.c:5508 Hangup
>>> sofia/internal/1006 at public.ip.removed [CS_EXECUTE] [NORMAL_UNSPECIFIED]
>>> 2011-06-22 13:33:50.514941 [DEBUG] switch_channel.c:2657 Send signal
>>> sofia/internal/1006 at public.ip.removed [KILL]
>>> 2011-06-22 13:33:50.514941 [DEBUG] switch_core_session.c:1118 Send signal
>>> sofia/internal/1006 at public.ip.removed [BREAK]
>>> 2011-06-22 13:33:50.534966 [DEBUG] switch_ivr_play_say.c:1649 done
>>> playing file
>>> 2011-06-22 13:33:50.625988 [DEBUG] switch_ivr_play_say.c:244 Handle
>>> play-file:[voicemail/vm-press.wav] (en:en)
>>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_session.c:2063
>>> sofia/internal/1006 at public.ip.removed skip receive message
>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:371 (
>>> sofia/internal/1006 at public.ip.removed) State EXECUTE going to sleep
>>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:325 (
>>> sofia/internal/1006 at public.ip.removed) Running State Change CS_HANGUP
>>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:565 (
>>> sofia/internal/1006 at public.ip.removed) State HANGUP
>>> 2011-06-22 13:33:50.727027 [DEBUG] mod_sofia.c:458 Channel
>>> sofia/internal/1006 at public.ip.removed hanging up, cause:
>>> NORMAL_UNSPECIFIED
>>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:46
>>> sofia/internal/1006 at public.ip.removed Standard HANGUP, cause:
>>> NORMAL_UNSPECIFIED
>>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:565 (
>>> sofia/internal/1006 at public.ip.removed) State HANGUP going to sleep
>>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:356 (
>>> sofia/internal/1006 at public.ip.removed) State Change CS_HANGUP ->
>>> CS_REPORTING
>>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_session.c:1118 Send signal
>>> sofia/internal/1006 at public.ip.removed [BREAK]
>>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:325 (
>>> sofia/internal/1006 at public.ip.removed) Running State Change CS_REPORTING
>>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:625 (
>>> sofia/internal/1006 at public.ip.removed) State REPORTING
>>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:53
>>> sofia/internal/1006 at public.ip.removed Standard REPORTING, cause:
>>> NORMAL_UNSPECIFIED
>>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:625 (
>>> sofia/internal/1006 at public.ip.removed) State REPORTING going to sleep
>>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:350 (
>>> sofia/internal/1006 at public.ip.removed) State Change CS_REPORTING ->
>>> CS_DESTROY
>>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_session.c:1118 Send signal
>>> sofia/internal/1006 at public.ip.removed [BREAK]
>>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_session.c:1290 Session 5 (
>>> sofia/internal/1006 at public.ip.removed) Locked, Waiting on external
>>> entities
>>> 2011-06-22 13:33:50.740064 [NOTICE] switch_core_session.c:1308 Session 5
>>> (sofia/internal/1006 at public.ip.removed) Ended
>>> 2011-06-22 13:33:50.740064 [NOTICE] switch_core_session.c:1310 Close
>>> Channel sofia/internal/1006 at public.ip.removed [CS_DESTROY]
>>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:454 (
>>> sofia/internal/1006 at public.ip.removed) Callstate Change HANGUP -> DOWN
>>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:457 (
>>> sofia/internal/1006 at public.ip.removed) Running State Change CS_DESTROY
>>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:467 (
>>> sofia/internal/1006 at public.ip.removed) State DESTROY
>>> 2011-06-22 13:33:50.740064 [DEBUG] mod_sofia.c:363
>>> sofia/internal/1006 at public.ip.removed SOFIA DESTROY
>>> 2011-06-22 13:33:50.780056 [DEBUG] switch_nat.c:570 unmapped public port
>>> 31484 protocol UDP to localport 31484
>>> 2011-06-22 13:33:50.840070 [DEBUG] switch_nat.c:570 unmapped public port
>>> 31485 protocol UDP to localport 31485
>>> 2011-06-22 13:33:50.840070 [DEBUG] switch_core_state_machine.c:60
>>> sofia/internal/1006 at public.ip.removed Standard DESTROY
>>> 2011-06-22 13:33:50.840070 [DEBUG] switch_core_state_machine.c:467 (
>>> sofia/internal/1006 at public.ip.removed) State DESTROY going to sleep
>>>
>>> The above example was from an externally situated SIP phone ringing
>>> voicemail (4000) on FreeSWITCH.
>>>
>>> I have experimented changing various timers and timeouts in the config of
>>> FreeSWITCH (one at a time, being careful to put them back afterwards!) but
>>> been unable to resolve the issue.
>>>
>>> Incidentally, we have no long term intention of running off-site SIP
>>> phones with the PBX and I'm hoping not to have to leave it in the DMZ
>>> either, it's just like that for debugging. What is a real issue is the calls
>>> to our external SIP provider (i.e. outbound calls) being dropped.
>>>
>>> Any suggestions would be greatly appreciated.
>>>
>>> Thanks,
>>>
>>> Matthew Ralston
>>> Web Developer & IT Consultant
>>>
>>> matt at mralston.co.uk
>>> www.mralston.com
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
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>>> FreeSWITCH-users at lists.freeswitch.org
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>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
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>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
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>>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
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>
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>
>
>
> _______________________________________________
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>
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From david.ponzone at ipeva.fr  Thu Jun 23 23:59:16 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 23 Jun 2011 21:59:16 +0200
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <4E037E03.2070408@statirasystems.com>
References: <4E0219B2.9070808@statirasystems.com>	<4E0227C8.9080304@statirasystems.com>	<4E02340D.2050708@statirasystems.com>	<BANLkTiksPsLhd23O-ws626svwnKcJJRmXw@mail.gmail.com>	<BANLkTinAjxG3Bp7jy3GTRs1km_DdoFDrCA@mail.gmail.com>	<BANLkTimip1BOfOG+U9rBOAJeiUJK_PL4fw@mail.gmail.com>	<BANLkTimBC4FKan=kjNoizbGUFGAe1ACu-A@mail.gmail.com>	<AC8EC5A4-1799-433A-9DD0-A715ADF5F256@freeswitch.org>	<BANLkTikQmkxi_C_1aQocH3O+nQrt4eaA9A@mail.gmail.com>	<B4D334F2-EE02-4EFB-A4A3-165C55CF7606@ipeva.fr>	<4E03669C.2030603@statirasystems.com>	<963C184F-7466-4874-80F1-0CC13FEC8581@ipeva.fr>	<4E036EE5.60208@statirasystems.com>	<4E036F3B.8090303@statirasystems.com>	<4E037099.9090107@statirasystems.com>	<4E0374B7.8030709@statirasystems.com>	<F2D1C5FE-6BF3-4537-AB64-F47004DBA73F@ipeva.fr>
	<F7B1B3E9-F6E4-4CF5-8A44-994E4EE1B902@ipeva.fr>
	<4E037E03.2070408@statirasystems.com>
Message-ID: <183C4D22-20BB-4695-AC7F-82AFE0D4D385@ipeva.fr>

I had the same feeling about blue.box.
I think there are more interesting GUIs out there, like FreePBX V3.
The weird thing is that none of them seems very active.

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

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

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




Le 23/06/2011 ? 19:55, William Moore a ?crit :

> I give up on blue.box. They seem preoccupied by there other projects. Documentation does not explain enough. It is like describing a human by their parts only and not how it interacts. Peace out.
> William J. Moore
> Statira Systems
> 611 Caroline St
> Fredericksburg, VA
> http://www.statirasystems.com
> 
> On 06/23/2011 01:34 PM, David Ponzone wrote:
>> 
>> In sipinterface_2, make the following modification:
>> 
>>  <param name="auth-calls" value="false"/> 
>> 
>> 
>>         
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> 
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> 
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>> 
>> 
>> 
>> 
>> Le 23/06/2011 ? 19:31, David Ponzone a ?crit :
>> 
>>> The issue is that you are expecting them to authenticate, but they won't.
>>> 
>>> David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>> 
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>> 
>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>> 
>>> 
>>> 
>>> 
>>> Le 23/06/2011 ? 19:15, William Moore a ?crit :
>>> 
>>>> Ok, I reregistered and called right away. I get the following:
>>>>   0.000000 66.227.100.20 -> 192.168.1.254 SIP/SDP Request: INVITE sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with session description
>>>>   0.000766 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>>>>   0.002321 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy Authentication Required
>>>>   0.021120 66.227.100.20 -> 192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>>>   3.272168 192.168.1.254 -> 66.227.100.20 SIP Request: REGISTER sip:jnctn.net;transport=udp
>>>>   3.292076 66.227.100.20 -> 192.168.1.254 SIP Status: 200 OK    (1 bindings)
>>>> 
>>>> The call just states it is a none working number. That means something in my dial plan is incorrect. I am double checking it now.
>>>> William J. Moore
>>>> Statira Systems
>>>> 611 Caroline St
>>>> Fredericksburg, VA
>>>> http://www.statirasystems.com
>>>> 
>>>> On 06/23/2011 12:58 PM, William Moore wrote:
>>>>> 
>>>>> When I put it in and it reloads, blue.box removes it. Can I reference the gateway in another sip interface file or does it have to be in that one?
>>>>> William J. Moore
>>>>> Statira Systems
>>>>> 611 Caroline St
>>>>> Fredericksburg, VA
>>>>> http://www.statirasystems.com
>>>>> 
>>>>> On 06/23/2011 12:52 PM, William Moore wrote:
>>>>>> 
>>>>>> correction: missed a line.
>>>>>> 
>>>>>>   0.000000 66.227.100.20 -> 192.168.1.254 SIP/SDP Request: INVITE sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with session description
>>>>>>   0.000814 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>>>>>>   0.002440 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy Authentication Required
>>>>>>   0.023173 66.227.100.20 -> 192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>>>>> 
>>>>>> William J. Moore
>>>>>> Statira Systems
>>>>>> 611 Caroline St
>>>>>> Fredericksburg, VA
>>>>>> http://www.statirasystems.com
>>>>>> 
>>>>>> On 06/23/2011 12:50 PM, William Moore wrote:
>>>>>>> 
>>>>>>> I get this with wireshark
>>>>>>> 
>>>>>>>  40.345679 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>>>>>>>  40.347381 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy Authentication Required
>>>>>>>  40.367906 66.227.100.20 -> 192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>>>>>> 
>>>>>>> I am looking for the expiration now..
>>>>>>> William J. Moore
>>>>>>> Statira Systems
>>>>>>> 611 Caroline St
>>>>>>> Fredericksburg, VA
>>>>>>> http://www.statirasystems.com
>>>>>>> 
>>>>>>> On 06/23/2011 12:25 PM, David Ponzone wrote:
>>>>>>>> 
>>>>>>>> Take a sip trace on your box with tcpdump or preferably, tshark:
>>>>>>>> tshark port 5080
>>>>>>>> 
>>>>>>>> You will then see if you receive it.
>>>>>>>> 
>>>>>>>> In the config you sent, I don't see anything about register expiration, or NAT keepalive, so I would really recommend you add a ping every 30 seconds to your gateway config.
>>>>>>>> 
>>>>>>>> If the issue is there, it's quite easy to see:
>>>>>>>> unregister the gateway
>>>>>>>> register it again
>>>>>>>> make an inbound call right away
>>>>>>>> 
>>>>>>>> David Ponzone  Direction Technique
>>>>>>>> email: david.ponzone at ipeva.fr
>>>>>>>> tel:      01 74 03 18 97
>>>>>>>> gsm:   06 66 98 76 34
>>>>>>>> 
>>>>>>>> Service Client IPeva
>>>>>>>> tel:      0811 46 26 26
>>>>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>>>>> 
>>>>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> Le 23/06/2011 ? 18:15, William Moore a ?crit :
>>>>>>>> 
>>>>>>>>> It was the 1 and it needs to be set for e.164(+1). Blue.box makes it a little more difficult since it wants to manage the xmls. I could have set it up as a separate config file however that defeats the purpose of the GUI.
>>>>>>>>> 
>>>>>>>>> I still have the issue of incoming calls not even showing up in sofia. Junction Networks says it is going out, but my server is not responding. I'm not sure if I have the settings for incoming correct.
>>>>>>>>> William J. Moore
>>>>>>>>> Statira Systems
>>>>>>>>> 611 Caroline St
>>>>>>>>> Fredericksburg, VA
>>>>>>>>> http://www.statirasystems.com
>>>>>>>>> 
>>>>>>>>> On 06/23/2011 04:21 AM, David Ponzone wrote:
>>>>>>>>>> 
>>>>>>>>>> William,
>>>>>>>>>> 
>>>>>>>>>> If I believe the example configuraiton on Junction's web, they don't expect E164:
>>>>>>>>>> 
>>>>>>>>>>     <extension name="junctionnetworks.com>
>>>>>>>>>>     <condition field="destination_number" expression="^(1{0,1}\d{10})$">
>>>>>>>>>>       <action application="set"data="effective_caller_id_number=OUTBOUNDCALLERID"/>
>>>>>>>>>>       <action application="bridge" data="sofia/gateway/junctionnetworks.com/$1"/>
>>>>>>>>>>      </condition>
>>>>>>>>>>    </extension>
>>>>>>>>>> But, elsewhere in their knowledgebase, they say:
>>>>>>>>>> 
>>>>>>>>>> Numbering
>>>>>>>>>> The standard e.164 numbering plan (ITU) is used. North American numbers are required to be prefixed with a '1'. International numbers need to be prefixed with "011".
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> I think they are confused about what is E164....
>>>>>>>>>> 
>>>>>>>>>> You should try to prefix with 1 as Michael recommended.
>>>>>>>>>> 
>>>>>>>>>> David Ponzone  Direction Technique
>>>>>>>>>> email: david.ponzone at ipeva.fr
>>>>>>>>>> tel:      01 74 03 18 97
>>>>>>>>>> gsm:   06 66 98 76 34
>>>>>>>>>> 
>>>>>>>>>> Service Client IPeva
>>>>>>>>>> tel:      0811 46 26 26
>>>>>>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>>>>>>> 
>>>>>>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> Le 23/06/2011 ? 08:19, Michael Collins a ?crit :
>>>>>>>>>> 
>>>>>>>>>>> Perhaps the gateway wants the leading 1 or expects e.164 format?
>>>>>>>>>>> -MC
>>>>>>>>>>> 
>>>>>>>>>>> On Wed, Jun 22, 2011 at 5:37 PM, Brian West <brian at freeswitch.org> wrote:
>>>>>>>>>>> 484 is address incomplete means you didn't send enough digits to the gateway usually.
>>>>>>>>>>> 
>>>>>>>>>>> /b
>>>>>>>>>>> 
>>>>>>>>>>> On Jun 22, 2011, at 7:34 PM, Brad Mina wrote:
>>>>>>>>>>> 
>>>>>>>>>>>> Ah, thought it was recv not send. Good to know that an invalid gateway issue
>>>>>>>>>>>> can cause this.
>>>>>>>>>>>> 
>>>>>>>>>>>> On Wed, Jun 22, 2011 at 5:31 PM, Michael Collins <msc at freeswitch.org> wrote:
>>>>>>>>>>>> 
>>>>>>>>>>>>> The 484 I believe is a result of the bad gateway. In any case, the OP needs
>>>>>>>>>>>>> to figure out what's up w/ that gw or no calls will be going out...
>>>>>>>>>>>>> 
>>>>>>>>>>>>> -MC
>>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>> 
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>> 
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>> 
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>> 
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>> 
>>>>>>>> 
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>> 
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>> 
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>> 
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From bmoore at statirasystems.com  Fri Jun 24 01:10:00 2011
From: bmoore at statirasystems.com (William Moore)
Date: Thu, 23 Jun 2011 17:10:00 -0400
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <183C4D22-20BB-4695-AC7F-82AFE0D4D385@ipeva.fr>
References: <4E0219B2.9070808@statirasystems.com>	<4E0227C8.9080304@statirasystems.com>	<4E02340D.2050708@statirasystems.com>	<BANLkTiksPsLhd23O-ws626svwnKcJJRmXw@mail.gmail.com>	<BANLkTinAjxG3Bp7jy3GTRs1km_DdoFDrCA@mail.gmail.com>	<BANLkTimip1BOfOG+U9rBOAJeiUJK_PL4fw@mail.gmail.com>	<BANLkTimBC4FKan=kjNoizbGUFGAe1ACu-A@mail.gmail.com>	<AC8EC5A4-1799-433A-9DD0-A715ADF5F256@freeswitch.org>	<BANLkTikQmkxi_C_1aQocH3O+nQrt4eaA9A@mail.gmail.com>	<B4D334F2-EE02-4EFB-A4A3-165C55CF7606@ipeva.fr>	<4E03669C.2030603@statirasystems.com>	<963C184F-7466-4874-80F1-0CC13FEC8581@ipeva.fr>	<4E036EE5.60208@statirasystems.com>	<4E036F3B.8090303@statirasystems.com>	<4E037099.9090107@statirasystems.com>	<4E0374B7.8030709@statirasystems.com>	<F2D1C5FE-6BF3-4537-AB64-F47004DBA73F@ipeva.fr>	<F7B1B3E9-F6E4-4CF5-8A44-994E4EE1B902@ipeva.fr>	<4E037E03.2070408@statirasystems.com>
	<183C4D22-20BB-4695-AC7F-82AFE0D4D385@ipeva.fr>
Message-ID: <4E03ABA8.4030204@statirasystems.com>

I noticed that also. I have been pondering doing my own. I can program 
but am fairly new to the freeswitch and sip in general.

Don't want anything all that fancy. Just working would be good.

William J. Moore
Statira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com


On 06/23/2011 03:59 PM, David Ponzone wrote:
> I had the same feeling about blue.box.
> I think there are more interesting GUIs out there, like FreePBX V3.
> The weird thing is that none of them seems very active.
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service ClientIPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
> diffusion non autoris?e est interdite. Tout message ?lectronique est 
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au 
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
> n'?tes pas destinataire de ce message, merci de le d?truire 
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 23/06/2011 ? 19:55, William Moore a ?crit :
>
>> I give up on blue.box. They seem preoccupied by there other projects. 
>> Documentation does not explain enough. It is like describing a human 
>> by their parts only and not how it interacts. Peace out.
>> William J. Moore
>> Statira Systems
>> 611 Caroline St
>> Fredericksburg, VA
>> http://www.statirasystems.com
>>
>> On 06/23/2011 01:34 PM, David Ponzone wrote:
>>> In sipinterface_2, make the following modification:
>>>
>>> <param name="auth-calls" value="false"/>
>>>
>>>
>>> David Ponzone Direction Technique
>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service ClientIPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
>>> <BLOCKED::http://www.ipeva-studio.com/>
>>>
>>> /Ce message et toutes les pi?ces jointes sont confidentiels et 
>>> ?tablis ? l'intention exclusive de ses destinataires. Toute 
>>> utilisation ou diffusion non autoris?e est interdite. Tout message 
>>> ?lectronique est susceptible d'alt?ration. /*/IPeva/*/ d?cline toute 
>>> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou 
>>> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le 
>>> d?truire imm?diatement et d'avertir l'exp?diteur./
>>> /
>>> /
>>>
>>>
>>>
>>> Le 23/06/2011 ? 19:31, David Ponzone a ?crit :
>>>
>>>> The issue is that you are expecting them to authenticate, but they 
>>>> won't.
>>>>
>>>> David Ponzone Direction Technique
>>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>>> tel:      01 74 03 18 97
>>>> gsm:   06 66 98 76 34
>>>>
>>>> Service ClientIPeva
>>>> tel:      0811 46 26 26
>>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - 
>>>> www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/>
>>>>
>>>> /Ce message et toutes les pi?ces jointes sont confidentiels et 
>>>> ?tablis ? l'intention exclusive de ses destinataires. Toute 
>>>> utilisation ou diffusion non autoris?e est interdite. Tout message 
>>>> ?lectronique est susceptible d'alt?ration. /*/IPeva/*/ d?cline 
>>>> toute responsabilit? au titre de ce message s'il a ?t? alt?r?, 
>>>> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, 
>>>> merci de le d?truire imm?diatement et d'avertir l'exp?diteur./
>>>> /
>>>> /
>>>>
>>>>
>>>>
>>>> Le 23/06/2011 ? 19:15, William Moore a ?crit :
>>>>
>>>>> Ok, I reregistered and called right away. I get the following:
>>>>>    0.000000 66.227.100.20 ->  192.168.1.254 SIP/SDP Request: INVITEsip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with session description
>>>>>    0.000766 192.168.1.254 ->  66.227.100.20 SIP Status: 100 Trying
>>>>>    0.002321 192.168.1.254 ->  66.227.100.20 SIP Status: 407 Proxy Authentication Required
>>>>>    0.021120 66.227.100.20 ->  192.168.1.254 SIP Request: ACKsip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>>>>    3.272168 192.168.1.254 ->  66.227.100.20 SIP Request: REGISTERsip:jnctn.net;transport=udp
>>>>>    3.292076 66.227.100.20 ->  192.168.1.254 SIP Status: 200 OK    (1 bindings)
>>>>>
>>>>> The call just states it is a none working number. That means something in my dial plan is incorrect. I am double checking it now.
>>>>> William J. Moore
>>>>> Statira Systems
>>>>> 611 Caroline St
>>>>> Fredericksburg, VA
>>>>> http://www.statirasystems.com
>>>>>
>>>>> On 06/23/2011 12:58 PM, William Moore wrote:
>>>>>> When I put it in and it reloads, blue.box removes it. Can I 
>>>>>> reference the gateway in another sip interface file or does it 
>>>>>> have to be in that one?
>>>>>> William J. Moore
>>>>>> Statira Systems
>>>>>> 611 Caroline St
>>>>>> Fredericksburg, VA
>>>>>> http://www.statirasystems.com
>>>>>>
>>>>>> On 06/23/2011 12:52 PM, William Moore wrote:
>>>>>>> correction: missed a line.
>>>>>>>
>>>>>>>   0.000000 66.227.100.20 -> 192.168.1.254 SIP/SDP Request: 
>>>>>>> INVITE 
>>>>>>> sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, 
>>>>>>> with session description
>>>>>>>   0.000814 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>>>>>>>   0.002440 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy 
>>>>>>> Authentication Required
>>>>>>>   0.023173 66.227.100.20 -> 192.168.1.254 SIP Request: ACK 
>>>>>>> sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>>>>>>
>>>>>>> William J. Moore
>>>>>>> Statira Systems
>>>>>>> 611 Caroline St
>>>>>>> Fredericksburg, VA
>>>>>>> http://www.statirasystems.com
>>>>>>>
>>>>>>> On 06/23/2011 12:50 PM, William Moore wrote:
>>>>>>>> I get this with wireshark
>>>>>>>>
>>>>>>>>  40.345679 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>>>>>>>>  40.347381 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy 
>>>>>>>> Authentication Required
>>>>>>>>  40.367906 66.227.100.20 -> 192.168.1.254 SIP Request: ACK 
>>>>>>>> sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>>>>>>>
>>>>>>>> I am looking for the expiration now..
>>>>>>>> William J. Moore
>>>>>>>> Statira Systems
>>>>>>>> 611 Caroline St
>>>>>>>> Fredericksburg, VA
>>>>>>>> http://www.statirasystems.com
>>>>>>>>
>>>>>>>> On 06/23/2011 12:25 PM, David Ponzone wrote:
>>>>>>>>> Take a sip trace on your box with tcpdump or preferably, tshark:
>>>>>>>>> tshark port 5080
>>>>>>>>>
>>>>>>>>> You will then see if you receive it.
>>>>>>>>>
>>>>>>>>> In the config you sent, I don't see anything about register 
>>>>>>>>> expiration, or NAT keepalive, so I would really recommend you 
>>>>>>>>> add a ping every 30 seconds to your gateway config.
>>>>>>>>>
>>>>>>>>> If the issue is there, it's quite easy to see:
>>>>>>>>> unregister the gateway
>>>>>>>>> register it again
>>>>>>>>> make an inbound call right away
>>>>>>>>>
>>>>>>>>> David Ponzone Direction Technique
>>>>>>>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>>>>>>>> tel:      01 74 03 18 97
>>>>>>>>> gsm:   06 66 98 76 34
>>>>>>>>>
>>>>>>>>> Service ClientIPeva
>>>>>>>>> tel:      0811 46 26 26
>>>>>>>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - 
>>>>>>>>> www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/>
>>>>>>>>>
>>>>>>>>> /Ce message et toutes les pi?ces jointes sont confidentiels et 
>>>>>>>>> ?tablis ? l'intention exclusive de ses destinataires. Toute 
>>>>>>>>> utilisation ou diffusion non autoris?e est interdite. Tout 
>>>>>>>>> message ?lectronique est susceptible d'alt?ration. 
>>>>>>>>> /*/IPeva/*/ d?cline toute responsabilit? au titre de ce 
>>>>>>>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes 
>>>>>>>>> pas destinataire de ce message, merci de le d?truire 
>>>>>>>>> imm?diatement et d'avertir l'exp?diteur./
>>>>>>>>> /
>>>>>>>>> /
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Le 23/06/2011 ? 18:15, William Moore a ?crit :
>>>>>>>>>
>>>>>>>>>> It was the 1 and it needs to be set for e.164(+1). Blue.box 
>>>>>>>>>> makes it a little more difficult since it wants to manage the 
>>>>>>>>>> xmls. I could have set it up as a separate config file 
>>>>>>>>>> however that defeats the purpose of the GUI.
>>>>>>>>>>
>>>>>>>>>> I still have the issue of incoming calls not even showing up 
>>>>>>>>>> in sofia. Junction Networks says it is going out, but my 
>>>>>>>>>> server is not responding. I'm not sure if I have the settings 
>>>>>>>>>> for incoming correct.
>>>>>>>>>> William J. Moore
>>>>>>>>>> Statira Systems
>>>>>>>>>> 611 Caroline St
>>>>>>>>>> Fredericksburg, VA
>>>>>>>>>> http://www.statirasystems.com
>>>>>>>>>>
>>>>>>>>>> On 06/23/2011 04:21 AM, David Ponzone wrote:
>>>>>>>>>>> William,
>>>>>>>>>>>
>>>>>>>>>>> If I believe the example configuraiton on Junction's web, 
>>>>>>>>>>> they don't expect E164:
>>>>>>>>>>>
>>>>>>>>>>>     <extension name="junctionnetworks.com  <http://junctionnetworks.com/>>
>>>>>>>>>>>      <condition field="destination_number" expression="^(1{0,1}\d{10})$">
>>>>>>>>>>>        <action application="set"data="effective_caller_id_number=OUTBOUNDCALLERID"/>
>>>>>>>>>>>        <action application="bridge" data="sofia/gateway/junctionnetworks.com/$1"/>
>>>>>>>>>>>       </condition>
>>>>>>>>>>>     </extension>
>>>>>>>>>>> But, elsewhere in their knowledgebase, they say:
>>>>>>>>>>>
>>>>>>>>>>> *Numbering*
>>>>>>>>>>> The standard e.164 numbering plan (ITU 
>>>>>>>>>>> <http://www.itu.int/>) is used. North American numbers are 
>>>>>>>>>>> required to be prefixed with a '1'. International numbers 
>>>>>>>>>>> need to be prefixed with "011".
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> I think they are confused about what is E164....
>>>>>>>>>>>
>>>>>>>>>>> You should try to prefix with 1 as Michael recommended.
>>>>>>>>>>>
>>>>>>>>>>> David Ponzone Direction Technique
>>>>>>>>>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>>>>>>>>>> tel:      01 74 03 18 97
>>>>>>>>>>> gsm:   06 66 98 76 34
>>>>>>>>>>>
>>>>>>>>>>> Service ClientIPeva
>>>>>>>>>>> tel:      0811 46 26 26
>>>>>>>>>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - 
>>>>>>>>>>> www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/>
>>>>>>>>>>>
>>>>>>>>>>> /Ce message et toutes les pi?ces jointes sont confidentiels 
>>>>>>>>>>> et ?tablis ? l'intention exclusive de ses destinataires. 
>>>>>>>>>>> Toute utilisation ou diffusion non autoris?e est interdite. 
>>>>>>>>>>> Tout message ?lectronique est susceptible d'alt?ration. 
>>>>>>>>>>> /*/IPeva/*/ d?cline toute responsabilit? au titre de ce 
>>>>>>>>>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
>>>>>>>>>>> n'?tes pas destinataire de ce message, merci de le d?truire 
>>>>>>>>>>> imm?diatement et d'avertir l'exp?diteur./
>>>>>>>>>>> /
>>>>>>>>>>> /
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> Le 23/06/2011 ? 08:19, Michael Collins a ?crit :
>>>>>>>>>>>
>>>>>>>>>>>> Perhaps the gateway wants the leading 1 or expects e.164 
>>>>>>>>>>>> format?
>>>>>>>>>>>> -MC
>>>>>>>>>>>>
>>>>>>>>>>>> On Wed, Jun 22, 2011 at 5:37 PM, Brian West 
>>>>>>>>>>>> <brian at freeswitch.org <mailto:brian at freeswitch.org>> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>     484 is address incomplete means you didn't send enough
>>>>>>>>>>>>     digits to the gateway usually.
>>>>>>>>>>>>
>>>>>>>>>>>>     /b
>>>>>>>>>>>>
>>>>>>>>>>>>     On Jun 22, 2011, at 7:34 PM, Brad Mina wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>>     Ah, thought it was recv not send. Good to know that an
>>>>>>>>>>>>>     invalid gateway issue
>>>>>>>>>>>>>     can cause this.
>>>>>>>>>>>>>
>>>>>>>>>>>>>     On Wed, Jun 22, 2011 at 5:31 PM, Michael Collins
>>>>>>>>>>>>>     <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>>>     The 484 I believe is a result of the bad gateway. In
>>>>>>>>>>>>>>     any case, the OP needs
>>>>>>>>>>>>>>     to figure out what's up w/ that gw or no calls will
>>>>>>>>>>>>>>     be going out...
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>     -MC
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>     _______________________________________________
>>>>>>>>>>>>     Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>>     http://www.cluecon.com <http://www.cluecon.com/>
>>>>>>>>>>>>     877-7-4ACLUE
>>>>>>>>>>>>
>>>>>>>>>>>>     FreeSWITCH-users mailing list
>>>>>>>>>>>>     FreeSWITCH-users at lists.freeswitch.org
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>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>> http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>>>>>>>>>>
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org 
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>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>> http://www.cluecon.com  877-7-4ACLUE
>>>>>>>>>>>
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>> http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>>>>>>>>
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org 
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>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> http://www.cluecon.com  877-7-4ACLUE
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
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>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com  877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
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>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
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>>>>>
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>>>>> FreeSWITCH-users at lists.freeswitch.org 
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>>>>
>>>
>>>
>>> _______________________________________________
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>> _______________________________________________
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>
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From gcd at i.ph  Fri Jun 24 02:38:55 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Fri, 24 Jun 2011 06:38:55 +0800
Subject: [Freeswitch-users] Call drops after 30 seconds
In-Reply-To: <BANLkTimBBN8RpAcEiJM6Rj6tegZSeydmUg@mail.gmail.com>
References: <71F1700A-7641-43E5-A7C6-F25A0DC2EBC5@mralston.com>
	<BANLkTikoP+hvtbzDn6fm_O6MmSN--x7TyA@mail.gmail.com>
	<C3BE3091-1685-42D1-ACD1-223B370607B7@mralston.com>
	<BANLkTi=4e1t607G0xCRygHGotxQrsXcnzw@mail.gmail.com>
	<28830F22-9654-45D2-B00A-1E69E32A42F9@mralston.com>
	<BANLkTimBBN8RpAcEiJM6Rj6tegZSeydmUg@mail.gmail.com>
Message-ID: <BANLkTikpEZHanBsTVmRTDLmtNq1HXx5Yag@mail.gmail.com>

matt,

i also encountered the same problem, even bridging internal endpoints. i
traced some RTCP packets using wireshark. i tried to enable the
"rtcp-audio-interval-msec=5000" in the internal sip profile. it solved the
problem momentarily.

-nandy

On Fri, Jun 24, 2011 at 2:32 AM, Michael Collins <msc at freeswitch.org> wrote:

> Matthew,
>
> Did you get resolution on this one? I think you can go to the sip profile
> and work with these lines:
>     <param name="ext-rtp-ip" value="auto-nat"/>
>     <param name="ext-sip-ip" value="auto-nat"/>
>
> You can put an IP address in those and try it out. Let us know what
> happens.
> -MC
>
> On Wed, Jun 22, 2011 at 8:26 AM, Matthew Ralston <freeswitch at mralston.com>wrote:
>
>> Hi Chris,
>>
>> The FreeSWITCH box does indeed have a private IP address. The Cisco
>> Linksys WAG160N router in front of it has UPnP enabled, not that I
>> particularly trust it.
>>
>> So maybe it's worth hard coding the public IP into the Sofia config.
>>
>> Where is the best place to put the public IP? The internal.xml and
>> external.xml files both refer to $${local_ip_v4} although I don't see where
>> this comes from. Does FreeSWITCH figure this out itself in some way?
>>
>> As I have internal SIP phones, an external SIP provider and (whilst
>> debugging) some external SIP phones (which use the internal profile!!). I'm
>> concerned that if I hard code the public IP address in to the wrong place it
>> will cause problems for the internal SIP phones.
>>
>>
>> Kind regards,
>>
>> Matthew Ralston
>> Web Developer & IT Consultant
>>
>> matt at mralston.co.uk
>> www.mralston.com
>>
>> On 22 Jun 2011, at 16:11, Chris Chen wrote:
>>
>> Hi Matthew, continue my last reply here
>>
>> 2) Is your FS server using private IP address?  you have to setup your FS
>> external SIP/RTP IP address to the proper public IP address, by either using
>> UPNP enabled router, STUN, or hardcoded public IP address in sofia profiles.
>>
>> Please check that.
>>
>> Thanks,
>> Chris
>>
>>
>> On Wed, Jun 22, 2011 at 10:49 AM, Matthew Ralston <
>> freeswitch at mralston.com> wrote:
>>
>>> Hi Chris,
>>>
>>> Thanks for the quick reply!
>>>
>>> The FreeSWITCH box is in the DMZ of a Cisco Linksys WAG160N. Best I can
>>> tell, the DMZ is doing its job and allowing all ports through, inbound and
>>> outbound, TCP & UDP.
>>>
>>> The external SIP phones are a Cisco SPA504G and Bria on the iPhone. These
>>> are behind behind a Cisco ASA5505, which has policy inspection for SIP
>>> switched on, i.e. ALG. I have also tested with Bria going over 3G (so it's
>>> not behind the Cisco ASA) and had the same problem.
>>>
>>> The other scenario we have is some Yealink T20P SIP phones on the same
>>> LAN segment as the FreeSWITCH box. These can make A-leg only calls into
>>> FreeSWITCH (like calling voicemail) and also calls to other internal SIP
>>> phones fine. However when they make an outbound call the problem happens
>>> again. In this case the b-leg of the calls are sent to an external SIP
>>> provider and get cut after 30 seconds. Incidentally, we also use the same
>>> SIP provider from an Asterisk box in our data centre and that doesn't have a
>>> problem, so I believe the SIP provider is fine.
>>>
>>> Kind regards,
>>>
>>>
>>> Matthew Ralston
>>> Web Developer & IT Consultant
>>>
>>> matt at mralston.co.uk
>>> www.mralston.com
>>>
>>> On 22 Jun 2011, at 14:21, Chris Chen wrote:
>>>
>>> Hi Matthew, this is typical behavior for the setup of SIP behind NAT.
>>>
>>>     1) Please provide the exact setup of remote SIP phones, what's the
>>> router model, does it have SIP ALG enabled, what kind  of SIP phones
>>>     2)
>>>
>>>
>>> On Wed, Jun 22, 2011 at 8:38 AM, Matthew Ralston <
>>> freeswitch at mralston.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> I'm having a problem at the moment with calls being successfully set up,
>>>> with two-way audio, being terminated by FreeSWITCH after 30 seconds.
>>>>
>>>> Internal calls (i.e. between SIP phones on the same LAN segment as the
>>>> FreeSWITCH box) work flawlessly.
>>>>
>>>> The problem arises when at least one of the handsets is located
>>>> elsewhere on the Internet. This behaviour is exhibited under the following
>>>> circumstances:
>>>>
>>>> - A-leg only call, e.g. to voicemail when the handset is at another
>>>> location on the Internet
>>>> - A-leg-B-leg call if one or both of the handsets are at another
>>>> location on the Internet
>>>> - Inbound calls from our external SIP provider
>>>> - Outbound calls to our external SIP provider
>>>>
>>>> So it is obvious that the problem is related to the SIP going via the
>>>> Internet, but I'm having trouble understanding why.
>>>>
>>>> Whilst debugging this problem I have placed the FreeSWITCH box is in the
>>>> DMZ on our router, so there should not be any ports blocked. The FreeSWITCH
>>>> box itself is not running a software firewall.
>>>>
>>>> The calls themselves are absolutely fine for the first 30 seconds - each
>>>> party can hear the other talking fine.
>>>>
>>>> The fact that the call is consistently dropped after 30 seconds (give or
>>>> take a second or two for PDD) suggests that some timeout is being triggered.
>>>>
>>>> When FreeSWITCH terminates the call, the following is logged to the
>>>> console:
>>>>
>>>> 2011-06-22 13:33:50.514941 [DEBUG] sofia.c:4787 Channel
>>>> sofia/internal/1006 at public.ip.removed entering state [terminating][0]
>>>> 2011-06-22 13:33:50.514941 [DEBUG] switch_channel.c:2641 (
>>>> sofia/internal/1006 at public.ip.removed) Callstate Change ACTIVE ->
>>>> HANGUP
>>>> 2011-06-22 13:33:50.514941 [NOTICE] sofia.c:5508 Hangup
>>>> sofia/internal/1006 at public.ip.removed [CS_EXECUTE] [NORMAL_UNSPECIFIED]
>>>> 2011-06-22 13:33:50.514941 [DEBUG] switch_channel.c:2657 Send signal
>>>> sofia/internal/1006 at public.ip.removed [KILL]
>>>> 2011-06-22 13:33:50.514941 [DEBUG] switch_core_session.c:1118 Send
>>>> signal sofia/internal/1006 at public.ip.removed [BREAK]
>>>> 2011-06-22 13:33:50.534966 [DEBUG] switch_ivr_play_say.c:1649 done
>>>> playing file
>>>> 2011-06-22 13:33:50.625988 [DEBUG] switch_ivr_play_say.c:244 Handle
>>>> play-file:[voicemail/vm-press.wav] (en:en)
>>>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_session.c:2063
>>>> sofia/internal/1006 at public.ip.removed skip receive message
>>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:371 (
>>>> sofia/internal/1006 at public.ip.removed) State EXECUTE going to sleep
>>>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:325 (
>>>> sofia/internal/1006 at public.ip.removed) Running State Change CS_HANGUP
>>>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:565 (
>>>> sofia/internal/1006 at public.ip.removed) State HANGUP
>>>> 2011-06-22 13:33:50.727027 [DEBUG] mod_sofia.c:458 Channel
>>>> sofia/internal/1006 at public.ip.removed hanging up, cause:
>>>> NORMAL_UNSPECIFIED
>>>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:46
>>>> sofia/internal/1006 at public.ip.removed Standard HANGUP, cause:
>>>> NORMAL_UNSPECIFIED
>>>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:565 (
>>>> sofia/internal/1006 at public.ip.removed) State HANGUP going to sleep
>>>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:356 (
>>>> sofia/internal/1006 at public.ip.removed) State Change CS_HANGUP ->
>>>> CS_REPORTING
>>>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_session.c:1118 Send
>>>> signal sofia/internal/1006 at public.ip.removed [BREAK]
>>>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:325 (
>>>> sofia/internal/1006 at public.ip.removed) Running State Change
>>>> CS_REPORTING
>>>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:625 (
>>>> sofia/internal/1006 at public.ip.removed) State REPORTING
>>>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:53
>>>> sofia/internal/1006 at public.ip.removed Standard REPORTING, cause:
>>>> NORMAL_UNSPECIFIED
>>>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:625 (
>>>> sofia/internal/1006 at public.ip.removed) State REPORTING going to sleep
>>>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:350 (
>>>> sofia/internal/1006 at public.ip.removed) State Change CS_REPORTING ->
>>>> CS_DESTROY
>>>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_session.c:1118 Send
>>>> signal sofia/internal/1006 at public.ip.removed [BREAK]
>>>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_session.c:1290 Session 5
>>>> (sofia/internal/1006 at public.ip.removed) Locked, Waiting on external
>>>> entities
>>>> 2011-06-22 13:33:50.740064 [NOTICE] switch_core_session.c:1308 Session 5
>>>> (sofia/internal/1006 at public.ip.removed) Ended
>>>> 2011-06-22 13:33:50.740064 [NOTICE] switch_core_session.c:1310 Close
>>>> Channel sofia/internal/1006 at public.ip.removed [CS_DESTROY]
>>>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:454 (
>>>> sofia/internal/1006 at public.ip.removed) Callstate Change HANGUP -> DOWN
>>>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:457 (
>>>> sofia/internal/1006 at public.ip.removed) Running State Change CS_DESTROY
>>>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:467 (
>>>> sofia/internal/1006 at public.ip.removed) State DESTROY
>>>> 2011-06-22 13:33:50.740064 [DEBUG] mod_sofia.c:363
>>>> sofia/internal/1006 at public.ip.removed SOFIA DESTROY
>>>> 2011-06-22 13:33:50.780056 [DEBUG] switch_nat.c:570 unmapped public port
>>>> 31484 protocol UDP to localport 31484
>>>> 2011-06-22 13:33:50.840070 [DEBUG] switch_nat.c:570 unmapped public port
>>>> 31485 protocol UDP to localport 31485
>>>> 2011-06-22 13:33:50.840070 [DEBUG] switch_core_state_machine.c:60
>>>> sofia/internal/1006 at public.ip.removed Standard DESTROY
>>>> 2011-06-22 13:33:50.840070 [DEBUG] switch_core_state_machine.c:467 (
>>>> sofia/internal/1006 at public.ip.removed) State DESTROY going to sleep
>>>>
>>>> The above example was from an externally situated SIP phone ringing
>>>> voicemail (4000) on FreeSWITCH.
>>>>
>>>> I have experimented changing various timers and timeouts in the config
>>>> of FreeSWITCH (one at a time, being careful to put them back afterwards!)
>>>> but been unable to resolve the issue.
>>>>
>>>> Incidentally, we have no long term intention of running off-site SIP
>>>> phones with the PBX and I'm hoping not to have to leave it in the DMZ
>>>> either, it's just like that for debugging. What is a real issue is the calls
>>>> to our external SIP provider (i.e. outbound calls) being dropped.
>>>>
>>>> Any suggestions would be greatly appreciated.
>>>>
>>>> Thanks,
>>>>
>>>> Matthew Ralston
>>>> Web Developer & IT Consultant
>>>>
>>>> matt at mralston.co.uk
>>>> www.mralston.com
>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
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>>>>
>>>
>>> _______________________________________________
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From msc at freeswitch.org  Fri Jun 24 03:07:38 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 23 Jun 2011 16:07:38 -0700
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <4E03ABA8.4030204@statirasystems.com>
References: <4E0219B2.9070808@statirasystems.com>
	<4E0227C8.9080304@statirasystems.com>
	<4E02340D.2050708@statirasystems.com>
	<BANLkTiksPsLhd23O-ws626svwnKcJJRmXw@mail.gmail.com>
	<BANLkTinAjxG3Bp7jy3GTRs1km_DdoFDrCA@mail.gmail.com>
	<BANLkTimip1BOfOG+U9rBOAJeiUJK_PL4fw@mail.gmail.com>
	<BANLkTimBC4FKan=kjNoizbGUFGAe1ACu-A@mail.gmail.com>
	<AC8EC5A4-1799-433A-9DD0-A715ADF5F256@freeswitch.org>
	<BANLkTikQmkxi_C_1aQocH3O+nQrt4eaA9A@mail.gmail.com>
	<B4D334F2-EE02-4EFB-A4A3-165C55CF7606@ipeva.fr>
	<4E03669C.2030603@statirasystems.com>
	<963C184F-7466-4874-80F1-0CC13FEC8581@ipeva.fr>
	<4E036EE5.60208@statirasystems.com>
	<4E036F3B.8090303@statirasystems.com>
	<4E037099.9090107@statirasystems.com>
	<4E0374B7.8030709@statirasystems.com>
	<F2D1C5FE-6BF3-4537-AB64-F47004DBA73F@ipeva.fr>
	<F7B1B3E9-F6E4-4CF5-8A44-994E4EE1B902@ipeva.fr>
	<4E037E03.2070408@statirasystems.com>
	<183C4D22-20BB-4695-AC7F-82AFE0D4D385@ipeva.fr>
	<4E03ABA8.4030204@statirasystems.com>
Message-ID: <BANLkTin3vWA=kxN6-jPM-k8Yfr7G25hZPQ@mail.gmail.com>

On Thu, Jun 23, 2011 at 2:10 PM, William Moore <bmoore at statirasystems.com>wrote:

> **
> I noticed that also. I have been pondering doing my own. I can program but
> am fairly new to the freeswitch and sip in general.
>
> Don't want anything all that fancy. Just working would be good.
>
If you want "just working, not fancy" then check out FusionPBX. It is a GUI
that is closer to the default FS XML configs than is blue.box. There are a
lot of people using it and it has a fair amount of traction, plus Mark Crane
(IRC: mcrane) is frequently in the #fusionpbx channel.

Whatever you do, please consider helping the existing projects before you
start a new one. Many have tried, few have succeeded... :)

-MC


> William J. Moore
> Statira Systems
> 611 Caroline St
> Fredericksburg, VAhttp://www.statirasystems.com
>
>
> On 06/23/2011 03:59 PM, David Ponzone wrote:
>
> I had the same feeling about blue.box.
> I think there are more interesting GUIs out there, like FreePBX V3.
> The weird thing is that none of them seems very active.
>
>   David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
>  Service Client IPeva
>  tel:      0811 46 26 26
>  www.ipeva.fr  -   www.ipeva-studio.com
>
>  *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
>  Le 23/06/2011 ? 19:55, William Moore a ?crit :
>
>  I give up on blue.box. They seem preoccupied by there other projects.
> Documentation does not explain enough. It is like describing a human by
> their parts only and not how it interacts. Peace out.
>
> William J. Moore
> Statira Systems
> 611 Caroline St
> Fredericksburg, VAhttp://www.statirasystems.com
>
>
> On 06/23/2011 01:34 PM, David Ponzone wrote:
>
> In sipinterface_2, make the following modification:
>
>   <param name="auth-calls" value="false"/>
>
>
>    David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
>  Service Client IPeva
>  tel:      0811 46 26 26
>  www.ipeva.fr  -   www.ipeva-studio.com
>
>  *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
>  Le 23/06/2011 ? 19:31, David Ponzone a ?crit :
>
>  The issue is that you are expecting them to authenticate, but they won't.
>
>
>   David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
>  Service Client IPeva
>  tel:      0811 46 26 26
>  www.ipeva.fr  -   www.ipeva-studio.com
>
>  *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
>  Le 23/06/2011 ? 19:15, William Moore a ?crit :
>
>  Ok, I reregistered and called right away. I get the following:
>
>   0.000000 66.227.100.20 -> 192.168.1.254 SIP/SDP Request: INVITE sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with session description
>   0.000766 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>   0.002321 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy Authentication Required
>   0.021120 66.227.100.20 -> 192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>   3.272168 192.168.1.254 -> 66.227.100.20 SIP Request: REGISTER sip:jnctn.net;transport=udp
>   3.292076 66.227.100.20 -> 192.168.1.254 SIP Status: 200 OK    (1 bindings)
>
> The call just states it is a none working number. That means something in my dial plan is incorrect. I am double checking it now.
>
> William J. Moore
> Statira Systems
> 611 Caroline St
> Fredericksburg, VAhttp://www.statirasystems.com
>
>
> On 06/23/2011 12:58 PM, William Moore wrote:
>
> When I put it in and it reloads, blue.box removes it. Can I reference the
> gateway in another sip interface file or does it have to be in that one?
>
> William J. Moore
> Statira Systems
> 611 Caroline St
> Fredericksburg, VAhttp://www.statirasystems.com
>
>
> On 06/23/2011 12:52 PM, William Moore wrote:
>
> correction: missed a line.
>
>   0.000000 66.227.100.20 -> 192.168.1.254 SIP/SDP Request: INVITE
> sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with session
> description
>   0.000814 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>   0.002440 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy
> Authentication Required
>   0.023173 66.227.100.20 -> 192.168.1.254 SIP Request: ACK
> sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>
> William J. Moore
> Statira Systems
> 611 Caroline St
> Fredericksburg, VAhttp://www.statirasystems.com
>
>
> On 06/23/2011 12:50 PM, William Moore wrote:
>
> I get this with wireshark
>
>  40.345679 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>  40.347381 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy
> Authentication Required
>  40.367906 66.227.100.20 -> 192.168.1.254 SIP Request: ACK
> sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>
> I am looking for the expiration now..
>
> William J. Moore
> Statira Systems
> 611 Caroline St
> Fredericksburg, VAhttp://www.statirasystems.com
>
>
> On 06/23/2011 12:25 PM, David Ponzone wrote:
>
> Take a sip trace on your box with tcpdump or preferably, tshark:
> tshark port 5080
>
> You will then see if you receive it.
>
>  In the config you sent, I don't see anything about register expiration,
> or NAT keepalive, so I would really recommend you add a ping every 30
> seconds to your gateway config.
>
>  If the issue is there, it's quite easy to see:
> unregister the gateway
> register it again
> make an inbound call right away
>
>   David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
>  Service Client IPeva
>  tel:      0811 46 26 26
>  www.ipeva.fr  -   www.ipeva-studio.com
>
>  *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
>  Le 23/06/2011 ? 18:15, William Moore a ?crit :
>
>  It was the 1 and it needs to be set for e.164(+1). Blue.box makes it a
> little more difficult since it wants to manage the xmls. I could have set it
> up as a separate config file however that defeats the purpose of the GUI.
>
> I still have the issue of incoming calls not even showing up in sofia.
> Junction Networks says it is going out, but my server is not responding. I'm
> not sure if I have the settings for incoming correct.
>
> William J. Moore
> Statira Systems
> 611 Caroline St
> Fredericksburg, VAhttp://www.statirasystems.com
>
>
> On 06/23/2011 04:21 AM, David Ponzone wrote:
>
> William,
>
>  If I believe the example configuraiton on Junction's web, they don't
> expect E164:
>
>     <extension name="junctionnetworks.com>
>     <condition field="destination_number" expression="^(1{0,1}\d{10})$">
>       <action application="set"data="effective_caller_id_number=OUTBOUNDCALLERID"/>
>       <action application="bridge" data="sofia/gateway/junctionnetworks.com/$1"/>
>      </condition>
>    </extension>
>
> But, elsewhere in their knowledgebase, they say:
>
>  *Numbering*
> The standard e.164 numbering plan (ITU <http://www.itu.int/>) is used.
> North American numbers are required to be prefixed with a '1'. International
> numbers need to be prefixed with "011".
>
>  I think they are confused about what is E164....
>
>  You should try to prefix with 1 as Michael recommended.
>
>    David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
>  Service Client IPeva
>  tel:      0811 46 26 26
>  www.ipeva.fr  -   www.ipeva-studio.com
>
>  *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
>  Le 23/06/2011 ? 08:19, Michael Collins a ?crit :
>
> Perhaps the gateway wants the leading 1 or expects e.164 format?
> -MC
>
> On Wed, Jun 22, 2011 at 5:37 PM, Brian West <brian at freeswitch.org> wrote:
>
>> 484 is address incomplete means you didn't send enough digits to the
>> gateway usually.
>>
>>  /b
>>
>>  On Jun 22, 2011, at 7:34 PM, Brad Mina wrote:
>>
>> Ah, thought it was recv not send. Good to know that an invalid gateway
>> issue
>> can cause this.
>>
>> On Wed, Jun 22, 2011 at 5:31 PM, Michael Collins <msc at freeswitch.org>
>> wrote:
>>
>> The 484 I believe is a result of the bad gateway. In any case, the OP
>> needs
>>
>> to figure out what's up w/ that gw or no calls will be going out...
>>
>>
>>  -MC
>>
>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
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>>
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From clive at lansink.co.nz  Fri Jun 24 03:21:05 2011
From: clive at lansink.co.nz (Clive Lansink)
Date: Fri, 24 Jun 2011 11:21:05 +1200
Subject: [Freeswitch-users] Recovering from database fault
Message-ID: <20110623232119.D65B5E8035@jlo.kiwilink.co.nz>

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From msc at freeswitch.org  Fri Jun 24 03:39:01 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 23 Jun 2011 16:39:01 -0700
Subject: [Freeswitch-users] Recovering from database fault
In-Reply-To: <20110623232119.D65B5E8035@jlo.kiwilink.co.nz>
References: <20110623232119.D65B5E8035@jlo.kiwilink.co.nz>
Message-ID: <BANLkTikiKPnB+Aj4aeMa974uCK3-AaYLYg@mail.gmail.com>

Which files in the db/ directory did you actually delete? There are several
files in there and generally it is "safe" to delete them all. Note that if
you delete the voicemail_xxx.db for your domain then you will lose user
voicemail information, like new vs. saved msgs. Of course, the recordings
themselves reside in storage/voicemail/xxx/. (xxx = the name of your domain,
so if it's the default configs then your domain is "default")

-MC

On Thu, Jun 23, 2011 at 4:21 PM, Clive Lansink <clive at lansink.co.nz> wrote:

> Hello list
>
> After a computer crash yesterday I noticed FreeSwitch reporting that the
> database disk image was malformed. After googling on this it seemed that the
> thing to do was to either delete or rename the db folder and get FreeSwitch
> to recreate it. After I did that and restarted Freeswitch, I found that all
> my phones seemed to register ok in that their web interfaces reported they
> were registered and I could make calls on them, but none of them could ring.
> When I rang another extension, it seemed from the log that the
> sofia/internal calling phone would get to the point where it would bridge to
> the sofia/internal for the extension, and then there would be an error
> something like could not create the channel, cause, user not registered. So
> all the phones could make calls but not receive them.
>
> I needed to get things working quickly and fortunately I achieved this by
> brute force - simply uninstalling Freeswitch, getting rid of its folder,
> installing again, and copying a saved copy of the conf directory back. Our
> installation is simple and we don't rely on the internal voicemail etc, so
> that was a quick way to get going again.
>
> But does anyone know what was going on here? Are there other data files
> apart from the db folder we should be aware of?
>
>
> Clive Lansink
> Email: Clive at Lansink.Co.NZ
> Phone: +64 9 520-4242
> Mobile: +64 21 663-999
> Fax: +64 21 789-150
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From cmcureau at gmail.com  Fri Jun 24 03:59:00 2011
From: cmcureau at gmail.com (Chris Cureau)
Date: Thu, 23 Jun 2011 18:59:00 -0500
Subject: [Freeswitch-users] (no subject)
Message-ID: <3491149370155049607@unknownmsgid>

I'm guessing this is just a pipe dream, but I'll throw it out anyway...

Is there any database available that can determine the carrier for an
outbound number? It would be really nice to be able to route cellular calls
via a cell phone automatically instead of setting up specific numbers in the
profile.
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From krice at freeswitch.org  Fri Jun 24 04:02:36 2011
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 23 Jun 2011 19:02:36 -0500
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <3491149370155049607@unknownmsgid>
Message-ID: <CA293E4C.18131%krice@freeswitch.org>

That?s called the LRN database and its not free... There are services that
will offer you access to it on a per query basis out there tho

K


On 6/23/11 6:59 PM, "Chris Cureau" <cmcureau at gmail.com> wrote:

> I'm guessing this is just a pipe dream, but I'll throw it out anyway...
> 
> Is there any database available that can determine the carrier for an outbound
> number? It would be really nice to be able to route cellular calls via a cell
> phone automatically instead of setting up specific numbers in the profile.
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From msc at freeswitch.org  Fri Jun 24 04:18:02 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 23 Jun 2011 17:18:02 -0700
Subject: [Freeswitch-users] Answer issue on inbound call
In-Reply-To: <BANLkTim3pTPV804t_Usk59eG2C56HAyFdA@mail.gmail.com>
References: <BANLkTin+f=uDMiOouyLWAorPvNTd1Z6xaA@mail.gmail.com>
	<BANLkTim=TOQD1tUvmHjeRuhAR-eKkGYgsw@mail.gmail.com>
	<BANLkTim3pTPV804t_Usk59eG2C56HAyFdA@mail.gmail.com>
Message-ID: <BANLkTi=Q3qRBCDRARDWp15=Guwyq-jtNLQ@mail.gmail.com>

I'm not sure I understand the problem. What is happening vs. what you
believe should be happening?
-MC

On Thu, Jun 23, 2011 at 3:31 AM, Max Alex <max.asterisk at gmail.com> wrote:

> Hi,
> Thanks for your reply.
> Here is my configuration and log
> http://pastebin.freeswitch.org/16571
>
> I am using A200 analog sangoma device with freeswitch, it is working fine
> but when it is routing call to 1001 then it is answered.
> Please provider your suggestions.
>
> Thanks,
> Max Alex
> Voip Developer
>
>
>
>
> On Wed, Jun 22, 2011 at 8:51 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> I thought the A200 was an analog card? Maybe I have my numbers mixed up...
>>
>> Go ahead and collect a debug log of this call. It might help to have your
>> configs posted as well. Use pastebin.freeswitch.org. See this wiki
>> article for tips on how to collect information:
>> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>
>> -MC
>>
>> On Wed, Jun 22, 2011 at 3:23 AM, Max Alex <max.asterisk at gmail.com> wrote:
>>
>>> Hi,
>>> I have installed freeswitch latest version with sangoma card A200 as
>>> well,
>>> I have installed and configured freetdm module with wanpipe drivers for
>>> freeswitch,
>>> We are properly receiving the inbound calls in public context and then we
>>> are routing that call to 1001 extension,
>>> it is properly routing to 1001 as well, but we have one issue while
>>> routing on 1001.
>>>
>>> Here is the issue description.
>>> I am calling from my cell phone to that DID number of pri line, and then
>>> it will start ringing on 1001 extension,
>>> When 1001 extension start ringing the call is answered on my cell phone,
>>> it is something like freeswitch preanswer or autoanswer the call, how can
>>> i stop this answer call when it is ringing on 1001 extension,
>>> Waiting for good reply.
>>>
>>> Thanks,
>>> Max Alex
>>>
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From d at d-man.org  Fri Jun 24 05:13:25 2011
From: d at d-man.org (Darren Schreiber)
Date: Thu, 23 Jun 2011 18:13:25 -0700
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <4E037E03.2070408@statirasystems.com>
Message-ID: <CA293246.2B652%d@d-man.org>

Hi William,
Sorry to hear that ? what exactly makes you think this? We're very active on the blue.box front. There have been 3 new features and over 20 bug fixes in just the past four weeks. Can you email me off-list to get whatever issue you're having sorted out?

Also, for authentication, bluebox "replaces" your changes because you're supposed to use the GUI to change that option. Go into Connectivity / SIP Interfaces and edit the SIP port you're trying to change. Uncheck the mark named "Require Authentication". It's not very difficult?

- Darren

--


From: William Moore <bmoore at statirasystems.com<mailto:bmoore at statirasystems.com>>
Reply-To: "freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>" <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Date: Thu, 23 Jun 2011 10:55:15 -0700
To: "freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>" <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] Trunk Issue

I give up on blue.box. They seem preoccupied by there other projects. Documentation does not explain enough. It is like describing a human by their parts only and not how it interacts. Peace out.

William J. Moore
Statira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com

On 06/23/2011 01:34 PM, David Ponzone wrote:
In sipinterface_2, make the following modification:

 <param name="auth-calls" value="false"/>

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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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




Le 23/06/2011 ? 19:31, David Ponzone a ?crit :

The issue is that you are expecting them to authenticate, but they won't.

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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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




Le 23/06/2011 ? 19:15, William Moore a ?crit :

Ok, I reregistered and called right away. I get the following:

  0.000000 66.227.100.20 -> 192.168.1.254 SIP/SDP Request: INVITE sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with session description
  0.000766 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
  0.002321 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy Authentication Required
  0.021120 66.227.100.20 -> 192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
  3.272168 192.168.1.254 -> 66.227.100.20 SIP Request: REGISTER sip:jnctn.net;transport=udp
  3.292076 66.227.100.20 -> 192.168.1.254 SIP Status: 200 OK    (1 bindings)

The call just states it is a none working number. That means something in my dial plan is incorrect. I am double checking it now.


William J. MooreStatira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com<http://www.statirasystems.com/>

On 06/23/2011 12:58 PM, William Moore wrote:
When I put it in and it reloads, blue.box removes it. Can I reference the gateway in another sip interface file or does it have to be in that one?

William J. Moore
Statira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com<http://www.statirasystems.com/>

On 06/23/2011 12:52 PM, William Moore wrote:
correction: missed a line.

  0.000000 66.227.100.20 -> 192.168.1.254 SIP/SDP Request: INVITE sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with session description
  0.000814 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
  0.002440 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy Authentication Required
  0.023173 66.227.100.20 -> 192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2


William J. Moore
Statira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com<http://www.statirasystems.com/>

On 06/23/2011 12:50 PM, William Moore wrote:
I get this with wireshark

 40.345679 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
 40.347381 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy Authentication Required
 40.367906 66.227.100.20 -> 192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2

I am looking for the expiration now..

William J. Moore
Statira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com<http://www.statirasystems.com/>

On 06/23/2011 12:25 PM, David Ponzone wrote:
Take a sip trace on your box with tcpdump or preferably, tshark:
tshark port 5080

You will then see if you receive it.

In the config you sent, I don't see anything about register expiration, or NAT keepalive, so I would really recommend you add a ping every 30 seconds to your gateway config.

If the issue is there, it's quite easy to see:
unregister the gateway
register it again
make an inbound call right away

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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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




Le 23/06/2011 ? 18:15, William Moore a ?crit :

It was the 1 and it needs to be set for e.164(+1). Blue.box makes it a little more difficult since it wants to manage the xmls. I could have set it up as a separate config file however that defeats the purpose of the GUI.

I still have the issue of incoming calls not even showing up in sofia. Junction Networks says it is going out, but my server is not responding. I'm not sure if I have the settings for incoming correct.

William J. Moore
Statira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com<http://www.statirasystems.com/>

On 06/23/2011 04:21 AM, David Ponzone wrote:
William,

If I believe the example configuraiton on Junction's web, they don't expect E164:


   <extension name="junctionnetworks.com<http://junctionnetworks.com/>>
    <condition field="destination_number" expression="^(1{0,1}\d{10})$">
      <action application="set"data="effective_caller_id_number=OUTBOUNDCALLERID"/>
      <action application="bridge" data="sofia/gateway/junctionnetworks.com/$1"/>
     </condition>
   </extension>


But, elsewhere in their knowledgebase, they say:


Numbering
The standard e.164 numbering plan (ITU<http://www.itu.int/>) is used. North American numbers are required to be prefixed with a '1'. International numbers need to be prefixed with "011".

I think they are confused about what is E164....

You should try to prefix with 1 as Michael recommended.

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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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




Le 23/06/2011 ? 08:19, Michael Collins a ?crit :

Perhaps the gateway wants the leading 1 or expects e.164 format?
-MC

On Wed, Jun 22, 2011 at 5:37 PM, Brian West <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:
484 is address incomplete means you didn't send enough digits to the gateway usually.

/b

On Jun 22, 2011, at 7:34 PM, Brad Mina wrote:

Ah, thought it was recv not send. Good to know that an invalid gateway issue
can cause this.

On Wed, Jun 22, 2011 at 5:31 PM, Michael Collins <msc at freeswitch.org<mailto:msc at freeswitch.org>> wrote:

The 484 I believe is a result of the bad gateway. In any case, the OP needs
to figure out what's up w/ that gw or no calls will be going out...

-MC



_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
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_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
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_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
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From d at d-man.org  Fri Jun 24 05:14:23 2011
From: d at d-man.org (Darren Schreiber)
Date: Thu, 23 Jun 2011 18:14:23 -0700
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <183C4D22-20BB-4695-AC7F-82AFE0D4D385@ipeva.fr>
Message-ID: <CA2932CA.2B657%d@d-man.org>

David,
FreePBX v3 is now called blue.box. You're not making any sense? Is this an attempt at creating FUD?

- Darren

--


From: David Ponzone <david.ponzone at ipeva.fr<mailto:david.ponzone at ipeva.fr>>
Reply-To: "freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>" <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Date: Thu, 23 Jun 2011 12:59:16 -0700
To: "freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>" <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] Trunk Issue

I had the same feeling about blue.box.
I think there are more interesting GUIs out there, like FreePBX V3.
The weird thing is that none of them seems very active.

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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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




Le 23/06/2011 ? 19:55, William Moore a ?crit :

I give up on blue.box. They seem preoccupied by there other projects. Documentation does not explain enough. It is like describing a human by their parts only and not how it interacts. Peace out.

William J. Moore
Statira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com<http://www.statirasystems.com/>

On 06/23/2011 01:34 PM, David Ponzone wrote:
In sipinterface_2, make the following modification:

 <param name="auth-calls" value="false"/>

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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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




Le 23/06/2011 ? 19:31, David Ponzone a ?crit :

The issue is that you are expecting them to authenticate, but they won't.

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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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




Le 23/06/2011 ? 19:15, William Moore a ?crit :

Ok, I reregistered and called right away. I get the following:

  0.000000 66.227.100.20 -> 192.168.1.254 SIP/SDP Request: INVITE sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with session description
  0.000766 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
  0.002321 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy Authentication Required
  0.021120 66.227.100.20 -> 192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
  3.272168 192.168.1.254 -> 66.227.100.20 SIP Request: REGISTER sip:jnctn.net;transport=udp
  3.292076 66.227.100.20 -> 192.168.1.254 SIP Status: 200 OK    (1 bindings)

The call just states it is a none working number. That means something in my dial plan is incorrect. I am double checking it now.


William J. MooreStatira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com<http://www.statirasystems.com/>

On 06/23/2011 12:58 PM, William Moore wrote:
When I put it in and it reloads, blue.box removes it. Can I reference the gateway in another sip interface file or does it have to be in that one?

William J. Moore
Statira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com<http://www.statirasystems.com/>

On 06/23/2011 12:52 PM, William Moore wrote:
correction: missed a line.

  0.000000 66.227.100.20 -> 192.168.1.254 SIP/SDP Request: INVITE sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with session description
  0.000814 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
  0.002440 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy Authentication Required
  0.023173 66.227.100.20 -> 192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2


William J. Moore
Statira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com<http://www.statirasystems.com/>

On 06/23/2011 12:50 PM, William Moore wrote:
I get this with wireshark

 40.345679 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
 40.347381 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy Authentication Required
 40.367906 66.227.100.20 -> 192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2

I am looking for the expiration now..

William J. Moore
Statira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com<http://www.statirasystems.com/>

On 06/23/2011 12:25 PM, David Ponzone wrote:
Take a sip trace on your box with tcpdump or preferably, tshark:
tshark port 5080

You will then see if you receive it.

In the config you sent, I don't see anything about register expiration, or NAT keepalive, so I would really recommend you add a ping every 30 seconds to your gateway config.

If the issue is there, it's quite easy to see:
unregister the gateway
register it again
make an inbound call right away

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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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




Le 23/06/2011 ? 18:15, William Moore a ?crit :

It was the 1 and it needs to be set for e.164(+1). Blue.box makes it a little more difficult since it wants to manage the xmls. I could have set it up as a separate config file however that defeats the purpose of the GUI.

I still have the issue of incoming calls not even showing up in sofia. Junction Networks says it is going out, but my server is not responding. I'm not sure if I have the settings for incoming correct.

William J. Moore
Statira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com<http://www.statirasystems.com/>

On 06/23/2011 04:21 AM, David Ponzone wrote:
William,

If I believe the example configuraiton on Junction's web, they don't expect E164:


   <extension name="junctionnetworks.com<http://junctionnetworks.com/>>
    <condition field="destination_number" expression="^(1{0,1}\d{10})$">
      <action application="set"data="effective_caller_id_number=OUTBOUNDCALLERID"/>
      <action application="bridge" data="sofia/gateway/junctionnetworks.com/$1"/>
     </condition>
   </extension>


But, elsewhere in their knowledgebase, they say:


Numbering
The standard e.164 numbering plan (ITU<http://www.itu.int/>) is used. North American numbers are required to be prefixed with a '1'. International numbers need to be prefixed with "011".

I think they are confused about what is E164....

You should try to prefix with 1 as Michael recommended.

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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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




Le 23/06/2011 ? 08:19, Michael Collins a ?crit :

Perhaps the gateway wants the leading 1 or expects e.164 format?
-MC

On Wed, Jun 22, 2011 at 5:37 PM, Brian West <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:
484 is address incomplete means you didn't send enough digits to the gateway usually.

/b

On Jun 22, 2011, at 7:34 PM, Brad Mina wrote:

Ah, thought it was recv not send. Good to know that an invalid gateway issue
can cause this.

On Wed, Jun 22, 2011 at 5:31 PM, Michael Collins <msc at freeswitch.org<mailto:msc at freeswitch.org>> wrote:

The 484 I believe is a result of the bad gateway. In any case, the OP needs
to figure out what's up w/ that gw or no calls will be going out...

-MC



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From d at d-man.org  Fri Jun 24 05:15:12 2011
From: d at d-man.org (Darren Schreiber)
Date: Thu, 23 Jun 2011 18:15:12 -0700
Subject: [Freeswitch-users] SRTP
In-Reply-To: <201106161740.18059.justlikeef@gmail.com>
Message-ID: <CA29331C.2B65D%d@d-man.org>

Hi Rob,
	Sorry for responding late on this. I'd be happy to help with this - seems
like a really good use of time. Want to give a trial run at this over the
weekend?

- Darren


-- 






On 6/16/11 2:40 PM, "Rob Hutton" <justlikeef at gmail.com> wrote:

>I am trying to get encryption working from within Bluebox, in the most
>"reasonably flexible" way possible. (So no, not the default dialpan, but
>I 
>missed the example so I will go back and look at it)
>
>So, one scenario I am thinking needs to be supported is where you have
>two 
>devices that are registered to the same user, one encrypted and one not.
>For 
>instance, a phone and a remote ringer.
>
>What I am looking for is the best way to stay as flexible as possible.
>It may 
>be a situation where you end up turning on encryption system wide if the
>devices support it, but that is overkill in a situation where there is a
>seperate voice and data VLAN unless there is a need for that level of
>security..
>
>It may be a situation where I need to offer both options and write two
>dialplan enries in the situation where the admin wants to enable it
>device by 
>device.
>
>BTW, I am also using my head to beat through getting TLS working on the
>front 
>end.  I would REALLY appreciate another set of eyes if you have time.
>
>http://jira.freeswitch.org/browse/FS-3346?page=com.atlassian.jira.plugin.s
>ystem.issuetabpanels:comment-
>tabpanel&focusedCommentId=24719#action_24719
>
>Thanks,
>Rob
>
>On Thursday 16 June 2011 14:01:41 Michael Collins wrote:
>> Are you working off of the default.xml dialplan file? If so, it has an
>> example condition already:
>> 
>>       <condition field="${sip_has_crypto}"
>> expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$"
>> break="never">
>> <action application="set" data="sip_secure_media=true"/>
>> <!-- Offer SRTP on outbound legs if we have it on inbound. -->
>> <!-- <action application="export" data="sip_secure_media=true"/> -->
>>       </condition>
>> 
>> What exactly are you checking on in your scenario? Most likely there is
>>an
>> elegant way to do it. Give us the plain language description of the
>>problem
>> you're addressing and the community will no doubt have good suggestions
>>for
>> you.
>> 
>> -MC
>> 
>> On Thu, Jun 16, 2011 at 10:22 AM, Rob Hutton <justlikeef at gmail.com>
>>wrote:
>> > Steven -
>> > 
>> > Thanks for the help here...
>> > 
>> > So there would have to be two dialplan entries for this number to work
>> > with either RTP or SRTP? (Maybe two devices registering to the same
>> > user?)
>> > 
>> > Would it make more since to do this in a more global manner higher up
>>in
>> > the
>> > dialplan in its own condition block?
>> > 
>> > On Thursday 16 June 2011 03:15:33 Steven Ayre wrote:
>> > >  {ERR} mod_dialplan_xml.c:110 Nexted conditions are not allowed
>> > > 
>> > > That's because it shouldn't be nested. It's not missing a /, and the
>> > > 1st Should have the /. The extra indendation shouldn't be there on
>>the
>> > > 2nd.
>> > > 
>> > > It should look like this:
>> > > 
>> > > <extension name="incoming-fxs">
>> > > 
>> > >      <condition field="destination_number" expression="^(202)$"/>
>> > >      <condition field="${sip_has_crypto}"
>> > > 
>> > > expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$"
>> > > break="never">
>> > > 
>> > >           <action application="set" data="sip_secure_media=true"/>
>> > >           <action application="bridge" data="openzap/1/1"/>
>> > >      
>> > >      </condition>
>> > > 
>> > > </extension>
>> > > 
>> > > The two conditions function as an AND, even though it's not nested.
>>FS
>> > > stops checking the extension as soon as it sees a condition that's
>> > > false (at least by default and in the above case), so if the
>> > > destination is not 202 it'll never get to the 2nd condition.
>> > > 
>> > > -Steve
>> > > 
>> > > On 16 June 2011 03:10, Rob Hutton <justlikeef at gmail.com> wrote:
>> > > > I think I have TLS and SRTP working at this point, but in the
>>docs it
>> > > > says to use the following template for the dialplan:
>> > > > 
>> > > > http://wiki.freeswitch.org/wiki/Secure_RTP:
>> > > >  <extension name="incoming-fxs">
>> > > >  
>> > > >    <condition field="destination_number" expression="^(202)$"/>
>> > > >    
>> > > >      <condition field="${sip_has_crypto}"
>> > > > 
>> > > > expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$"
>> > > > break="never">
>> > > > 
>> > > >      <action application="set" data="sip_secure_media=true"/>
>> > > >      <action application="bridge" data="openzap/1/1"/>
>> > > >    
>> > > >    </condition>
>> > > >  
>> > > >  </extension
>> > > > 
>> > > > 1) There is a missing > at the end of the close extension tag.
>> > > > 2) There is either a missing / at the end of the internal
>>condition
>> > 
>> > line,
>> > 
>> > > > or a missing condition close tag somewhere
>> > > > 3) When I fix the interal condition, I get an error:
>> > > > 
>> > > > {ERR} mod_dialplan_xml.c:110 Nexted conditions are not allowed
>> > > > 
>> > > > All this, but a packet capture shows that SRTP is working based on
>> > > > what
>> > 
>> > I
>> > 
>> > > > did on:
>> > > > 
>> > > > http://wiki.freeswitch.org/wiki/SIP_TLS
>> > > > 
>> > > > Can someone give me some guidance on  the Secure_RTP page and I
>>will
>> > > > update whatever?
>> > > > 
>> > > > _______________________________________________
>> > > > FreeSWITCH-users mailing list
>> > > > FreeSWITCH-users at lists.freeswitch.org
>> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > 
>> > > > UNSUBSCRIBE:
>> > http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > 
>> > > > http://www.freeswitch.org
>> > 
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > 
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>
>_______________________________________________
>Join us at ClueCon 2011, Aug 9-11, Chicago
>http://www.cluecon.com 877-7-4ACLUE
>
>FreeSWITCH-users mailing list
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From d at d-man.org  Fri Jun 24 05:16:22 2011
From: d at d-man.org (Darren Schreiber)
Date: Thu, 23 Jun 2011 18:16:22 -0700
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <4E02340D.2050708@statirasystems.com>
Message-ID: <CA293365.2B661%d@d-man.org>

You've created a loop ? it means you've setup auth or the numbering incorrectly. Again, can we take a look at your box? This is pretty easy to solve?
--


From: William Moore <bmoore at statirasystems.com<mailto:bmoore at statirasystems.com>>
Reply-To: "freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>" <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Date: Wed, 22 Jun 2011 11:27:25 -0700
To: "freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>" <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] Trunk Issue

And what sophiatrace reports from my soft phone device 502 calling number 7032203446. When I call from outside in nothing is shown on the trace, so I am not sure what that means.

=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2011.06.22 14:22:40 =~=~=~=~=~=~=~=~=~=~=~=
send 695 bytes to udp/[192.168.1.11]:42615 at 20:28:07.684442:
   ------------------------------------------------------------------------
   OPTIONS sip:502 at 192.168.1.11:42615;ob<mailto:sip:502 at 192.168.1.11:42615;ob> SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.254;rport;branch=z9hG4bK4Hyp1U0Djc12F
   Max-Forwards: 70
   From: <sip:mod_sofia at 192.168.1.254:5060><mailto:sip:mod_sofia at 192.168.1.254:5060>;tag=trm8SvQyg57ga
   To: <sip:502 at phone.statirasystems.com><mailto:sip:502 at phone.statirasystems.com>
   Call-ID: fb2b0eb4-17b0-122f-85bc-0002a5ed0817
   CSeq: 14049563 OPTIONS
   Contact: <sip:mod_sofia at 192.168.1.254:5060><mailto:sip:mod_sofia at 192.168.1.254:5060>
   User-Agent: Configured by 2600hz!
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: precondition, path, replaces
   Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0

   ------------------------------------------------------------------------
recv 1188 bytes from udp/[192.168.1.11]:42615 at 20:28:07.755356:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 192.168.1.254;rport=5060;received=192.168.1.254;branch=z9hG4bK4Hyp1U0Djc12F
   Call-ID: fb2b0eb4-17b0-122f-85bc-0002a5ed0817
   From: <sip:mod_sofia at 192.168.1.254><mailto:sip:mod_sofia at 192.168.1.254>;tag=trm8SvQyg57ga
   To: <sip:502 at phone.statirasystems.com><mailto:sip:502 at phone.statirasystems.com>;tag=z9hG4bK4Hyp1U0Djc12F
   CSeq: 14049563 OPTIONS
   Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
   Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
   Supported: replaces, 100rel, timer, norefersub
   Allow-Events: presence, message-summary, refer
   User-Agent: CSipSimple r944 / inc-8
   Content-Type: application/sdp
   Content-Length:   427

   v=0
   o=- 3517756265 3517756265 IN IP4 192.168.1.11
   s=pjmedia
   c=IN IP4 192.168.1.11
   t=0 0
   m=audio 49493 RTP/AVP 9 104 103 105 102 0 8 101
   a=rtcp:40021 IN IP4 192.168.1.11
   a=rtpmap:9 G722/8000
   a=rtpmap:104 speex/32000
   a=rtpmap:103 speex/16000
   a=rtpmap:105 iLBC/8000
   a=fmtp:105 mode=30
   a=rtpmap:102 speex/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=sendrecv
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   ------------------------------------------------------------------------
2011-06-22 13:28:15.458709 [DEBUG] switch_nat.c:510 mapped public port 5060 protocol UDP to localport 5060
2011-06-22 13:28:15.574709 [DEBUG] switch_nat.c:510 mapped public port 5060 protocol TCP to localport 5060
2011-06-22 13:28:15.691704 [DEBUG] switch_nat.c:510 mapped public port 5070 protocol UDP to localport 5070
2011-06-22 13:28:15.808694 [DEBUG] switch_nat.c:510 mapped public port 5070 protocol TCP to localport 5070
2011-06-22 13:28:15.925687 [DEBUG] switch_nat.c:510 mapped public port 5080 protocol UDP to localport 5080
2011-06-22 13:28:16.040679 [DEBUG] switch_nat.c:510 mapped public port 5080 protocol TCP to localport 5080
recv 1346 bytes from udp/[192.168.1.11]:42615 at 20:28:31.536924:
   ------------------------------------------------------------------------
   INVITE sip:7032203446 at phone.statirasystems.com<mailto:sip:7032203446 at phone.statirasystems.com> SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.11:42615;rport;branch=z9hG4bKPjYU1YgUOgEnTrbjl09SB0FLBTUw3q.4kD
   Max-Forwards: 70
   From: <sip:502 at phone.statirasystems.com><mailto:sip:502 at phone.statirasystems.com>;tag=llr4Rj-HnTn9EQhAB8mLWxBLOUkqnap.
   To: <sip:7032203446 at phone.statirasystems.com><mailto:sip:7032203446 at phone.statirasystems.com>
   Contact: <sip:502 at 192.168.1.11:42615;ob><mailto:sip:502 at 192.168.1.11:42615;ob>;+sip.ice
   Call-ID: hiSXYJajRDw17Q9BBhV8MiVL6FHepRwl
   CSeq: 23484 INVITE
   Route: <sip:phone.statirasystems.com;transport=udp;lr>
   Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
   Supported: replaces, 100rel, timer, norefersub
   Session-Expires: 1800
   Min-SE: 90
   User-Agent: CSipSimple r944 / inc-8
   Content-Type: application/sdp
   Content-Length:   616

   v=0
   o=- 3517756289 3517756289 IN IP4 192.168.1.11
   s=pjmedia
   c=IN IP4 192.168.1.11
   t=0 0
   a=X-nat:0
   m=audio 33616 RTP/AVP 9 104 103 105 102 0 8 101
   a=rtcp:38327 IN IP4 192.168.1.11
   a=rtpmap:9 G722/8000
   a=rtpmap:104 speex/32000
   a=rtpmap:103 speex/16000
   a=rtpmap:105 iLBC/8000
   a=fmtp:105 mode=30
   a=rtpmap:102 speex/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=sendrecv
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=ice-ufrag:7adc4b5a
   a=ice-pwd:15961ee1
   a=candidate:Hc0a8010b 1 UDP 2130706431 192.168.1.11 33616 typ host
   a=candidate:Hc0a8010b 2 UDP 2130706430 192.168.1.11 38327 typ host
   ------------------------------------------------------------------------
send 363 bytes to udp/[192.168.1.11]:42615 at 20:28:31.537870:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 192.168.1.11:42615;rport=42615;branch=z9hG4bKPjYU1YgUOgEnTrbjl09SB0FLBTUw3q.4kD
   From: <sip:502 at phone.statirasystems.com><mailto:sip:502 at phone.statirasystems.com>;tag=llr4Rj-HnTn9EQhAB8mLWxBLOUkqnap.
   To: <sip:7032203446 at phone.statirasystems.com><mailto:sip:7032203446 at phone.statirasystems.com>
   Call-ID: hiSXYJajRDw17Q9BBhV8MiVL6FHepRwl
   CSeq: 23484 INVITE
   User-Agent: Configured by 2600hz!
   Content-Length: 0

   ------------------------------------------------------------------------
2011-06-22 13:28:31.539663 [DEBUG] sofia.c:6551 IP 192.168.1.11 Rejected by acl "net_list_5". Falling back to Digest auth.
2011-06-22 13:28:31.539663 [WARNING] sofia_reg.c:1246 SIP auth challenge (INVITE) on sofia profile 'sipinterface_1' for [7032203446 at phone.statirasystems.com<mailto:7032203446 at phone.statirasystems.com>] from ip 192.168.1.11
send 855 bytes to udp/[192.168.1.11]:42615 at 20:28:31.541775:
   ------------------------------------------------------------------------
   SIP/2.0 407 Proxy Authentication Required
   Via: SIP/2.0/UDP 192.168.1.11:42615;rport=42615;branch=z9hG4bKPjYU1YgUOgEnTrbjl09SB0FLBTUw3q.4kD
   From: <sip:502 at phone.statirasystems.com><mailto:sip:502 at phone.statirasystems.com>;tag=llr4Rj-HnTn9EQhAB8mLWxBLOUkqnap.
   To: <sip:7032203446 at phone.statirasystems.com><mailto:sip:7032203446 at phone.statirasystems.com>;tag=U1D1UQ81Dey3N
   Call-ID: hiSXYJajRDw17Q9BBhV8MiVL6FHepRwl
   CSeq: 23484 INVITE
   User-Agent: Configured by 2600hz!
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: precondition, path, replaces
   Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Proxy-Authenticate: Digest realm="phone.statirasystems.com", nonce="430ea594-c447-42e5-a795-1c5ac884b996", algorithm=MD5, qop="auth"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 445 bytes from udp/[192.168.1.11]:42615 at 20:28:31.544036:
   ------------------------------------------------------------------------
   ACK sip:7032203446 at phone.statirasystems.com<mailto:sip:7032203446 at phone.statirasystems.com> SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.11:42615;rport;branch=z9hG4bKPjYU1YgUOgEnTrbjl09SB0FLBTUw3q.4kD
   Max-Forwards: 70
   From: <sip:502 at phone.statirasystems.com><mailto:sip:502 at phone.statirasystems.com>;tag=llr4Rj-HnTn9EQhAB8mLWxBLOUkqnap.
   To: <sip:7032203446 at phone.statirasystems.com><mailto:sip:7032203446 at phone.statirasystems.com>;tag=U1D1UQ81Dey3N
   Call-ID: hiSXYJajRDw17Q9BBhV8MiVL6FHepRwl
   CSeq: 23484 ACK
   Route: <sip:phone.statirasystems.com;transport=udp;lr>
   Content-Length:  0

   ------------------------------------------------------------------------
recv 1643 bytes from udp/[192.168.1.11]:42615 at 20:28:31.545117:
   ------------------------------------------------------------------------
   INVITE sip:7032203446 at phone.statirasystems.com<mailto:sip:7032203446 at phone.statirasystems.com> SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.11:42615;rport;branch=z9hG4bKPjRaqtNZWIkhvyivUL2g2vkoVMTmdtxGlf
   Max-Forwards: 70
   From: <sip:502 at phone.statirasystems.com><mailto:sip:502 at phone.statirasystems.com>;tag=llr4Rj-HnTn9EQhAB8mLWxBLOUkqnap.
   To: <sip:7032203446 at phone.statirasystems.com><mailto:sip:7032203446 at phone.statirasystems.com>
   Contact: <sip:502 at 192.168.1.11:42615;ob><mailto:sip:502 at 192.168.1.11:42615;ob>;+sip.ice
   Call-ID: hiSXYJajRDw17Q9BBhV8MiVL6FHepRwl
   CSeq: 23485 INVITE
   Route: <sip:phone.statirasystems.com;transport=udp;lr>
   Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
   Supported: replaces, 100rel, timer, norefersub
   Session-Expires: 1800
   Min-SE: 90
   User-Agent: CSipSimple r944 / inc-8
   Proxy-Authorization: Digest username="502", realm="phone.statirasystems.com", nonce="430ea594-c447-42e5-a795-1c5ac884b996", uri="sip:7032203446 at phone.statirasystems.com"<mailto:sip:7032203446 at phone.statirasystems.com>, response="95d75cc4f1478333731f960e0e686399", algorithm=MD5, cnonce="VIYZuITTZROoTJla3ithxFWRKkW15K3H", qop=auth, nc=00000001
   Content-Type: application/sdp
   Content-Length:   616

   v=0
   o=- 3517756289 3517756289 IN IP4 192.168.1.11
   s=pjmedia
   c=IN IP4 192.168.1.11
   t=0 0
   a=X-nat:0
   m=audio 33616 RTP/AVP 9 104 103 105 102 0 8 101
   a=rtcp:38327 IN IP4 192.168.1.11
   a=rtpmap:9 G722/8000
   a=rtpmap:104 speex/32000
   a=rtpmap:103 speex/16000
   a=rtpmap:105 iLBC/8000
   a=fmtp:105 mode=30
   a=rtpmap:102 speex/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=sendrecv
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=ice-ufrag:7adc4b5a
   a=ice-pwd:15961ee1
   a=candidate:Hc0a8010b 1 UDP 2130706431 192.168.1.11 33616 typ host
   a=candidate:Hc0a8010b 2 UDP 2130706430 192.168.1.11 38327 typ host
   ------------------------------------------------------------------------
send 363 bytes to udp/[192.168.1.11]:42615 at 20:28:31.545859:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 192.168.1.11:42615;rport=42615;branch=z9hG4bKPjRaqtNZWIkhvyivUL2g2vkoVMTmdtxGlf
   From: <sip:502 at phone.statirasystems.com><mailto:sip:502 at phone.statirasystems.com>;tag=llr4Rj-HnTn9EQhAB8mLWxBLOUkqnap.
   To: <sip:7032203446 at phone.statirasystems.com><mailto:sip:7032203446 at phone.statirasystems.com>
   Call-ID: hiSXYJajRDw17Q9BBhV8MiVL6FHepRwl
   CSeq: 23485 INVITE
   User-Agent: Configured by 2600hz!
   Content-Length: 0

   ------------------------------------------------------------------------
2011-06-22 13:28:31.547460 [DEBUG] sofia.c:6551 IP 192.168.1.11 Rejected by acl "net_list_5". Falling back to Digest auth.
2011-06-22 13:28:31.547460 [NOTICE] switch_channel.c:816 New Channel sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> [ae454abd-dc4a-42c9-90cc-6b0b332c44f4]
2011-06-22 13:28:31.547460 [DEBUG] sofia.c:4761 Channel sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> entering state [received][100]
2011-06-22 13:28:31.547460 [DEBUG] sofia.c:4772 Remote SDP:
v=0

o=- 3517756289 3517756289 IN IP4 192.168.1.11

s=pjmedia

c=IN IP4 192.168.1.11

t=0 0

a=X-nat:0

m=audio 33616 RTP/AVP 9 104 103 105 102 0 8 101

a=rtpmap:9 G722/8000

a=rtpmap:104 speex/32000

a=rtpmap:103 speex/16000

a=rtpmap:105 iLBC/8000

a=fmtp:105 mode=30

a=rtpmap:102 speex/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=rtcp:38327 IN IP4 192.168.1.11

a=ice-ufrag:7adc4b5a

a=ice-pwd:15961ee1

a=candidate:Hc0a8010b 1 UDP 2130706431 192.168.1.11 33616 typ host

a=candidate:Hc0a8010b 2 UDP 2130706430 192.168.1.11 38327 typ host


2011-06-22 13:28:31.547460 [DEBUG] sofia_glue.c:4669 Audio Codec Compare [G722:9:8000:20:64000]/[G7221:115:32000:20:48000]
2011-06-22 13:28:31.547460 [DEBUG] sofia_glue.c:4669 Audio Codec Compare [G722:9:8000:20:64000]/[G7221:107:16000:20:32000]
2011-06-22 13:28:31.547460 [DEBUG] sofia_glue.c:4669 Audio Codec Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000]
2011-06-22 13:28:31.547460 [DEBUG] sofia_glue.c:2792 Set Codec sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> G722/8000 20 ms 160 samples 64000 bits
2011-06-22 13:28:31.547460 [DEBUG] sofia_glue.c:4783 Set 2833 dtmf send/recv payload to 101
2011-06-22 13:28:31.547460 [DEBUG] sofia.c:4943 (sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com>) State Change CS_NEW -> CS_INIT
2011-06-22 13:28:31.547460 [DEBUG] switch_core_session.c:1116 Send signal sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> [BREAK]
2011-06-22 13:28:31.553607 [DEBUG] switch_core_state_machine.c:325 (sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com>) Running State Change CS_INIT
2011-06-22 13:28:31.553607 [DEBUG] switch_core_state_machine.c:361 (sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com>) State INIT
2011-06-22 13:28:31.553607 [DEBUG] mod_sofia.c:84 sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> SOFIA INIT
2011-06-22 13:28:31.553607 [DEBUG] mod_sofia.c:124 (sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com>) State Change CS_INIT -> CS_ROUTING
2011-06-22 13:28:31.553607 [DEBUG] switch_core_session.c:1116 Send signal sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> [BREAK]
2011-06-22 13:28:31.553607 [DEBUG] switch_core_state_machine.c:361 (sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com>) State INIT going to sleep
2011-06-22 13:28:31.553607 [DEBUG] switch_core_state_machine.c:325 (sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com>) Running State Change CS_ROUTING
2011-06-22 13:28:31.553607 [DEBUG] switch_channel.c:1672 (sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com>) Callstate Change DOWN -> RINGING
2011-06-22 13:28:31.555041 [DEBUG] switch_core_state_machine.c:364 (sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com>) State ROUTING
2011-06-22 13:28:31.555041 [DEBUG] mod_sofia.c:147 sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> SOFIA ROUTING
2011-06-22 13:28:31.555041 [DEBUG] switch_core_state_machine.c:77 sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> Standard ROUTING
2011-06-22 13:28:31.555041 [INFO] mod_dialplan_xml.c:331 Processing 502 <502>->7032203446 in context context_1
Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> parsing [context_1->conditioning_callerid] continue=true
Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> Regex (PASS) [conditioning_callerid] ${internal_caller_id_number}(7036526678) =~ /^.+$/ break=on-false
Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> Action set(effective_caller_id_name=${internal_caller_id_name})
Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> Action set(effective_caller_id_number=${internal_caller_id_number})
Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> parsing [context_1->postroute_global] continue=true
Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> Absolute Condition [postroute_global]
Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> Action hash(insert/${domain_name}-last_dial/global/${uuid})
Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> parsing [context_1->main_number_1] continue=true
Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> Regex (FAIL) [main_number_1] destination_number(7032203446) =~ /^2001$/ break=on-false
Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> parsing [context_1->main_number_3] continue=true
Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> Regex (FAIL) [main_number_3] destination_number(7032203446) =~ /^502$/ break=on-false
Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> parsing [context_1->main_trunk_1_pattern_1] continue=true
Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> Regex (PASS) [main_trunk_1_pattern_1] destination_number(7032203446) =~ /^1{0,1}([2-9][0-8][0-9][2-9][0-9]{6})$/ break=never
Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> Action set(prepend=)
Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> Action set(effective_caller_id_name=Statira Systems)
Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> Action set(effective_caller_id_number=7036526678)
Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> Regex (PASS) [main_trunk_1_pattern_1] ${outbound_caller_id_number}(7036526678) =~ /^.+$/ break=never
Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> Action set(effective_caller_id_name=${outbound_caller_id_name})
Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> Action set(effective_caller_id_number=${outbound_caller_id_number})
Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> Action export(sip_cid_type=rpid)
Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> Regex (PASS) [main_trunk_1_pattern_1] destination_number(7032203446) =~ /^1{0,1}([2-9][0-8][0-9][2-9][0-9]{6})$/ break=never
Dialplan: sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> Action bridge(sofia/gateway/trunk_1/${prepend}7032203446)
2011-06-22 13:28:31.558679 [DEBUG] switch_core_state_machine.c:119 (sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com>) State Change CS_ROUTING -> CS_EXECUTE
2011-06-22 13:28:31.558679 [DEBUG] switch_core_session.c:1116 Send signal sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> [BREAK]
2011-06-22 13:28:31.558679 [DEBUG] switch_core_state_machine.c:364 (sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com>) State ROUTING going to sleep
2011-06-22 13:28:31.558679 [DEBUG] switch_core_state_machine.c:325 (sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com>) Running State Change CS_EXECUTE
2011-06-22 13:28:31.558679 [DEBUG] switch_core_state_machine.c:371 (sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com>) State EXECUTE
2011-06-22 13:28:31.558679 [DEBUG] mod_sofia.c:240 sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> SOFIA EXECUTE
2011-06-22 13:28:31.558679 [DEBUG] switch_core_state_machine.c:157 sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> Standard EXECUTE
EXECUTE sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> set(effective_caller_id_name=Monkey)
2011-06-22 13:28:31.559681 [DEBUG] mod_dptools.c:1060 sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> SET [effective_caller_id_name]=[Monkey]
EXECUTE sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> set(effective_caller_id_number=7036526678)
2011-06-22 13:28:31.559681 [DEBUG] mod_dptools.c:1060 sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> SET [effective_caller_id_number]=[7036526678]
EXECUTE sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> hash(insert/phone.statirasystems.com-spymap/502/ae454abd-dc4a-42c9-90cc-6b0b332c44f4)
EXECUTE sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> hash(insert/phone.statirasystems.com-last_dial/502/7032203446)
EXECUTE sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> hash(insert/phone.statirasystems.com-last_dial/global/ae454abd-dc4a-42c9-90cc-6b0b332c44f4)
EXECUTE sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> set(RFC2822_DATE=Wed, 22 Jun 2011 13:28:31 -0700)
2011-06-22 13:28:31.562678 [DEBUG] mod_dptools.c:1060 sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> SET [RFC2822_DATE]=[Wed, 22 Jun 2011 13:28:31 -0700]
EXECUTE sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> set(prepend=)
2011-06-22 13:28:31.563814 [DEBUG] mod_dptools.c:1060 sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> SET [prepend]=[UNDEF]
EXECUTE sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> set(effective_caller_id_name=Statira Systems)
2011-06-22 13:28:31.564681 [DEBUG] mod_dptools.c:1060 sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> SET [effective_caller_id_name]=[Statira Systems]
EXECUTE sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> set(effective_caller_id_number=7036526678)
2011-06-22 13:28:31.565679 [DEBUG] mod_dptools.c:1060 sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> SET [effective_caller_id_number]=[7036526678]
EXECUTE sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> set(effective_caller_id_name=Monkey)
2011-06-22 13:28:31.565679 [DEBUG] mod_dptools.c:1060 sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> SET [effective_caller_id_name]=[Monkey]
EXECUTE sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> set(effective_caller_id_number=7036526678)
2011-06-22 13:28:31.566804 [DEBUG] mod_dptools.c:1060 sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> SET [effective_caller_id_number]=[7036526678]
EXECUTE sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> export(sip_cid_type=rpid)
2011-06-22 13:28:31.567678 [DEBUG] switch_channel.c:965 EXPORT (export_vars) [sip_cid_type]=[rpid]
EXECUTE sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> bridge(sofia/gateway/trunk_1/7032203446)
2011-06-22 13:28:31.568678 [DEBUG] switch_channel.c:922 sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> EXPORTING[export_vars] [sip_cid_type]=[rpid] to event
2011-06-22 13:28:31.568678 [DEBUG] switch_ivr_originate.c:1873 Parsing global variables
2011-06-22 13:28:31.568678 [ERR] mod_sofia.c:4045 Invalid Gateway
2011-06-22 13:28:31.568678 [NOTICE] mod_sofia.c:4402 Close Channel N/A [CS_NEW]
2011-06-22 13:28:31.568678 [DEBUG] switch_core_state_machine.c:457 () Running State Change CS_DESTROY
2011-06-22 13:28:31.569833 [DEBUG] switch_core_state_machine.c:467 (N/A) State DESTROY
2011-06-22 13:28:31.569833 [DEBUG] mod_sofia.c:362 N/A SOFIA DESTROY
2011-06-22 13:28:31.569833 [DEBUG] switch_core_state_machine.c:467 (N/A) State DESTROY going to sleep
2011-06-22 13:28:31.569833 [ERR] switch_ivr_originate.c:2447 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
2011-06-22 13:28:31.569833 [DEBUG] switch_ivr_originate.c:3299 Originate Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]
2011-06-22 13:28:31.569833 [INFO] mod_dptools.c:2647 Originate Failed.  Cause: INVALID_NUMBER_FORMAT
2011-06-22 13:28:31.569833 [DEBUG] switch_channel.c:2567 (sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com>) Callstate Change RINGING -> HANGUP
2011-06-22 13:28:31.570693 [NOTICE] mod_dptools.c:2761 Hangup sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> [CS_EXECUTE] [INVALID_NUMBER_FORMAT]
2011-06-22 13:28:31.570693 [DEBUG] switch_channel.c:2583 Send signal sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> [KILL]
2011-06-22 13:28:31.570693 [DEBUG] switch_core_session.c:1116 Send signal sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> [BREAK]
2011-06-22 13:28:31.571679 [DEBUG] switch_core_session.c:2060 sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2011-06-22 13:28:31.571679 [DEBUG] switch_core_state_machine.c:371 (sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com>) State EXECUTE going to sleep
2011-06-22 13:28:31.571679 [DEBUG] switch_core_state_machine.c:325 (sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com>) Running State Change CS_HANGUP
2011-06-22 13:28:31.572811 [DEBUG] switch_core_state_machine.c:565 (sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com>) State HANGUP
2011-06-22 13:28:31.572811 [DEBUG] mod_sofia.c:457 Channel sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> hanging up, cause: INVALID_NUMBER_FORMAT
2011-06-22 13:28:31.573682 [DEBUG] mod_sofia.c:519 Responding to INVITE with: 484
send 861 bytes to udp/[192.168.1.11]:42615 at 20:28:31.574787:
   ------------------------------------------------------------------------
   SIP/2.0 484 Address Incomplete
   Via: SIP/2.0/UDP 192.168.1.11:42615;rport=42615;branch=z9hG4bKPjRaqtNZWIkhvyivUL2g2vkoVMTmdtxGlf
   From: <sip:502 at phone.statirasystems.com><mailto:sip:502 at phone.statirasystems.com>;tag=llr4Rj-HnTn9EQhAB8mLWxBLOUkqnap.
   To: <sip:7032203446 at phone.statirasystems.com><mailto:sip:7032203446 at phone.statirasystems.com>;tag=va7SXjS5aQmpH
   Call-ID: hiSXYJajRDw17Q9BBhV8MiVL6FHepRwl
   CSeq: 23485 INVITE
   User-Agent: Configured by 2600hz!
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: precondition, path, replaces
   Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Reason: Q.850;cause=28;text="INVALID_NUMBER_FORMAT"
   Content-Length: 0
   Remote-Party-ID: "7032203446" <sip:7032203446 at 192.168.1.254><mailto:sip:7032203446 at 192.168.1.254>;party=calling;privacy=off;screen=no

   ------------------------------------------------------------------------
2011-06-22 13:28:31.575063 [DEBUG] switch_core_state_machine.c:46 sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> Standard HANGUP, cause: INVALID_NUMBER_FORMAT
2011-06-22 13:28:31.575063 [DEBUG] switch_core_state_machine.c:565 (sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com>) State HANGUP going to sleep
2011-06-22 13:28:31.575063 [DEBUG] switch_core_state_machine.c:356 (sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com>) State Change CS_HANGUP -> CS_REPORTING
2011-06-22 13:28:31.575063 [DEBUG] switch_core_session.c:1116 Send signal sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> [BREAK]
2011-06-22 13:28:31.575063 [DEBUG] switch_core_state_machine.c:325 (sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com>) Running State Change CS_REPORTING
2011-06-22 13:28:31.575845 [DEBUG] switch_core_state_machine.c:625 (sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com>) State REPORTING
2011-06-22 13:28:31.575845 [DEBUG] switch_core_state_machine.c:53 sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> Standard REPORTING, cause: INVALID_NUMBER_FORMAT
recv 445 bytes from udp/[192.168.1.11]:42615 at 20:28:31.583298:
   ------------------------------------------------------------------------
   ACK sip:7032203446 at phone.statirasystems.com<mailto:sip:7032203446 at phone.statirasystems.com> SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.11:42615;rport;branch=z9hG4bKPjRaqtNZWIkhvyivUL2g2vkoVMTmdtxGlf
   Max-Forwards: 70
   From: <sip:502 at phone.statirasystems.com><mailto:sip:502 at phone.statirasystems.com>;tag=llr4Rj-HnTn9EQhAB8mLWxBLOUkqnap.
   To: <sip:7032203446 at phone.statirasystems.com><mailto:sip:7032203446 at phone.statirasystems.com>;tag=va7SXjS5aQmpH
   Call-ID: hiSXYJajRDw17Q9BBhV8MiVL6FHepRwl
   CSeq: 23485 ACK
   Route: <sip:phone.statirasystems.com;transport=udp;lr>
   Content-Length:  0

   ------------------------------------------------------------------------
2011-06-22 13:28:31.584480 [DEBUG] switch_core_state_machine.c:625 (sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com>) State REPORTING going to sleep
2011-06-22 13:28:31.585691 [DEBUG] switch_core_state_machine.c:350 (sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com>) State Change CS_REPORTING -> CS_DESTROY
2011-06-22 13:28:31.585691 [DEBUG] switch_core_session.c:1116 Send signal sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> [BREAK]
2011-06-22 13:28:31.585691 [DEBUG] switch_core_session.c:1288 Session 8 (sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com>) Locked, Waiting on external entities
2011-06-22 13:28:31.585691 [NOTICE] switch_core_session.c:1306 Session 8 (sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com>) Ended
2011-06-22 13:28:31.585691 [NOTICE] switch_core_session.c:1308 Close Channel sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> [CS_DESTROY]
2011-06-22 13:28:31.585691 [DEBUG] switch_core_state_machine.c:454 (sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com>) Callstate Change HANGUP -> DOWN
2011-06-22 13:28:31.586867 [DEBUG] switch_core_state_machine.c:457 (sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com>) Running State Change CS_DESTROY
2011-06-22 13:28:31.586867 [DEBUG] switch_core_state_machine.c:467 (sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com>) State DESTROY
2011-06-22 13:28:31.586867 [DEBUG] mod_sofia.c:362 sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> SOFIA DESTROY
2011-06-22 13:28:31.586867 [DEBUG] switch_core_state_machine.c:60 sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com> Standard DESTROY
2011-06-22 13:28:31.586867 [DEBUG] switch_core_state_machine.c:467 (sofia/sipinterface_1/502 at phone.statirasystems.com<mailto:sofia/sipinterface_1/502 at phone.statirasystems.com>) State DESTROY going to sleep
send 695 bytes to udp/[192.168.1.11]:42615 at 20:28:37.744017:
   ------------------------------------------------------------------------
   OPTIONS sip:502 at 192.168.1.11:42615;ob<mailto:sip:502 at 192.168.1.11:42615;ob> SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.254;rport;branch=z9hG4bK5tQF3pHHFNQNB
   Max-Forwards: 70
   From: <sip:mod_sofia at 192.168.1.254:5060><mailto:sip:mod_sofia at 192.168.1.254:5060>;tag=XK0jZDa97Za9c
   To: <sip:502 at phone.statirasystems.com><mailto:sip:502 at phone.statirasystems.com>
   Call-ID: 0d15c90c-17b1-122f-85bc-0002a5ed0817
   CSeq: 14049564 OPTIONS
   Contact: <sip:mod_sofia at 192.168.1.254:5060><mailto:sip:mod_sofia at 192.168.1.254:5060>
   User-Agent: Configured by 2600hz!
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: precondition, path, replaces
   Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0

   ------------------------------------------------------------------------
recv 1188 bytes from udp/[192.168.1.11]:42615 at 20:28:37.860253:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 192.168.1.254;rport=5060;received=192.168.1.254;branch=z9hG4bK5tQF3pHHFNQNB
   Call-ID: 0d15c90c-17b1-122f-85bc-0002a5ed0817
   From: <sip:mod_sofia at 192.168.1.254><mailto:sip:mod_sofia at 192.168.1.254>;tag=XK0jZDa97Za9c
   To: <sip:502 at phone.statirasystems.com><mailto:sip:502 at phone.statirasystems.com>;tag=z9hG4bK5tQF3pHHFNQNB
   CSeq: 14049564 OPTIONS
   Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
   Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
   Supported: replaces, 100rel, timer, norefersub
   Allow-Events: presence, message-summary, refer
   User-Agent: CSipSimple r944 / inc-8
   Content-Type: application/sdp
   Content-Length:   427

   v=0
   o=- 3517756295 3517756295 IN IP4 192.168.1.11
   s=pjmedia
   c=IN IP4 192.168.1.11
   t=0 0
   m=audio 49493 RTP/AVP 9 104 103 105 102 0 8 101
   a=rtcp:40021 IN IP4 192.168.1.11
   a=rtpmap:9 G722/8000
   a=rtpmap:104 speex/32000
   a=rtpmap:103 speex/16000
   a=rtpmap:105 iLBC/8000
   a=fmtp:105 mode=30
   a=rtpmap:102 speex/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=sendrecv
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   ------------------------------------------------------------------------


William J. Moore
Owner
Statira Systems
611 Caroline St
Fredericksburg, VA 22401
540.693.0579
www.statirasystems.com<http://www.statirasystems.com>

On 6/22/2011 1:35 PM, William Moore wrote:
Some details

/opt/freeswitch/conf/sip_profiles/bluebox_sipinterfaces.xml
<?xml version="1.0"?>
<include>
  <profile name="sipinterface_3">
    <settings>
      <param name="enable-timer" value="false"/>
      <param name="user-agent-string" value="Configured by 2600hz!"/>
      <param name="rtp-timer-name" value="soft"/>
      <param name="codec-prefs" value="$${global_codec_prefs}"/>
      <param name="inbound-codec-negotiation" value="generous"/>
      <param name="inbound-reg-force-matching-username" value="true"/>
      <param name="nonce-ttl" value="86400"/>
      <param name="rfc2833-pt" value="101"/>
      <param name="manage-presence" value="true"/>
      <param name="auth-calls" value="false"/>
      <param name="sip-ip" value="$${local_ip_v4}"/>
      <param name="rtp-ip" value="$${local_ip_v4}"/>
      <param name="sip-port" value="5070"/>
      <param name="ext-sip-ip" value="auto-nat"/>
      <param name="ext-rtp-ip" value="auto-nat"/>
      <param name="context" value="multitenant_routing_context"/>
      <param name="apply-nat-acl" value="nat.auto"/>
    </settings>
  </profile>
  <profile name="sipinterface_1">
    <settings>
      <param name="enable-timer" value="false"/>
      <param name="user-agent-string" value="Configured by 2600hz!"/>
      <param name="rtp-timer-name" value="soft"/>
      <param name="codec-prefs" value="$${global_codec_prefs}"/>
      <param name="inbound-codec-negotiation" value="generous"/>
      <param name="inbound-reg-force-matching-username" value="true"/>
      <param name="nonce-ttl" value="86400"/>
      <param name="rfc2833-pt" value="101"/>
      <param name="manage-presence" value="true"/>
      <param name="auth-calls" value="true"/>
      <param name="sip-ip" value="$${local_ip_v4}"/>
      <param name="rtp-ip" value="$${local_ip_v4}"/>
      <param name="sip-port" value="5060"/>
      <param name="nat-options-ping" value="true"/>
      <param name="all-reg-options-ping" value="true"/>
      <param name="ext-sip-ip" value="auto-nat"/>
      <param name="ext-rtp-ip" value="auto-nat"/>
      <param name="context" value="multitenant_routing_context"/>
      <param name="apply-nat-acl" value="nat.auto"/>
      <param name="apply-inbound-acl" value="net_list_5"/>
      <param name="aggressive-nat-detection" value="true"/>
      <param name="log-auth-failures" value="true"/>
    </settings>
    <gateways/>
  </profile>
  <profile name="sipinterface_2">
    <settings>
      <param name="enable-timer" value="false"/>
      <param name="user-agent-string" value="Configured by 2600hz!"/>
      <param name="rtp-timer-name" value="soft"/>
      <param name="codec-prefs" value="$${global_codec_prefs}"/>
      <param name="inbound-codec-negotiation" value="generous"/>
      <param name="inbound-reg-force-matching-username" value="true"/>
      <param name="nonce-ttl" value="86400"/>
      <param name="rfc2833-pt" value="101"/>
      <param name="manage-presence" value="true"/>
      <param name="auth-calls" value="true"/>
      <param name="sip-ip" value="$${local_ip_v4}"/>
      <param name="rtp-ip" value="$${local_ip_v4}"/>
      <param name="sip-port" value="5080"/>
      <param name="nat-options-ping" value="true"/>
      <param name="all-reg-options-ping" value="true"/>
      <param name="ext-sip-ip" value="auto-nat"/>
      <param name="ext-rtp-ip" value="auto-nat"/>
      <param name="context" value="multitenant_routing_context"/>
      <param name="apply-inbound-acl" value="net_list_5"/>
      <param name="apply-register-acl" value="rfc1918.auto"/>
      <param name="log-auth-failures" value="true"/>
      <param name="aggressive-nat-detection" value="true"/>
    </settings>
    <aliases>
      <alias name="$${location_1}"/>
    </aliases>
    <gateways>
      <gateway name="trunk_1">
        <param name="realm" value="jnctn.net"/>
        <param name="register-proxy" value="sip.jnctn.net"/>
        <param name="outbound-proxy" value="sip.jnctn.net"/>
        <param name="username" value="mowcow"/>
        <param name="password" value="RQcLt9ypz5DqoLz2"/>
        <param name="register" value="true"/>
      </gateway>
    </gateways>
    <domains>
      <domain name="all" alias="true" parse="false"/>
    </domains>
  </profile>
</include>


/opt/freeswitch/conf/dialplan/bluebox_dialplan.xml
<?xml version="1.0"?>
<include>
  <context name="context_1">
    <extension name="conditioning_callerid" continue="true">
      <condition field="${internal_caller_id_number}" expression="^.+$">
        <action application="set" data="effective_caller_id_name=${internal_caller_id_name}"/>
        <action application="set" data="effective_caller_id_number=${internal_caller_id_number}"/>
      </condition>
    </extension>
    <extension name="postroute_global" continue="true">
      <condition>
        <action application="hash" data="insert/${domain_name}-spymap/${caller_id_number}/${uuid}"/>
        <action application="hash" data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/>
        <action application="hash" data="insert/${domain_name}-last_dial/global/${uuid}"/>
        <action application="set" data="RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}"/>
      </condition>
    </extension>
    <extension name="main_number_1" continue="true">
      <condition field="destination_number" expression="^2001$">
        <action application="set" bluebox="settingEndBridge" data="hangup_after_bridge=true"/>
        <action application="set" bluebox="settingFail" data="continue_on_fail=true"/>
        <action application="set" bluebox="autoPlay" data="vm_auto_play=false"/>
        <action application="set" bluebox="settingTimeout" data="call_timeout=20"/>
        <action application="set" bluebox="settingRing" data="ringback=${us-ring}"/>
        <action application="set" bluebox="settingRingXfer" data="transfer_ringback=${us-ring}"/>
        <action application="export" bluebox="sipCalleeIdName" data="sip_callee_id_name=Account Admin"/>
        <action application="export" bluebox="sipCalleeIdNumber" data="sip_callee_id_number=2001"/>
        <action application="bridge" data="user/account_admin@$${location_1}"/>
        <action application="answer"/>
        <action application="voicemail" data="default voicemail_1 2001"/>
        <action application="hangup"/>
      </condition>
    </extension>
    <extension name="main_number_3" continue="true">
      <condition field="destination_number" expression="^502$">
        <action application="set" bluebox="settingEndBridge" data="hangup_after_bridge=true"/>
        <action application="set" bluebox="settingFail" data="continue_on_fail=true"/>
        <action application="set" bluebox="autoPlay" data="vm_auto_play=false"/>
        <action application="set" bluebox="settingTimeout" data="call_timeout=30"/>
        <action application="set" bluebox="settingRing" data="ringback=${us-ring}"/>
        <action application="set" bluebox="settingRingXfer" data="transfer_ringback=${us-ring}"/>
        <action application="export" bluebox="sipCalleeIdName" data="sip_callee_id_name=Monkey"/>
        <action application="export" bluebox="sipCalleeIdNumber" data="sip_callee_id_number=502"/>
        <action application="bridge" data="user/502@$${location_1}"/>
        <action application="answer"/>
        <action application="voicemail" data="default voicemail_1 2001"/>
        <action application="hangup"/>
      </condition>
    </extension>
    <extension name="main_trunk_1_pattern_1" continue="true">
      <condition field="destination_number" break="never" expression="^1{0,1}([2-9][0-8][0-9][2-9][0-9]{6})$" bluebox="pattern_1_part_1">
        <action application="set" bluebox="prepend" data="prepend="/>
        <action application="set" bluebox="cid_name" data="effective_caller_id_name=Statira Systems"/>
        <action application="set" bluebox="cid_number" data="effective_caller_id_number=7036526678"/>
      </condition>
      <condition field="${outbound_caller_id_number}" expression="^.+$" break="never" bluebox="caller_id">
        <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>
        <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
        <action application="export" bluebox="caller_id_field" data="sip_cid_type=rpid"/>
      </condition>
      <condition field="destination_number" break="never" expression="^1{0,1}([2-9][0-8][0-9][2-9][0-9]{6})$" bluebox="pattern_1_part_1_out">
        <action application="bridge" bluebox="out_trunk_1" data="sofia/gateway/trunk_1/${prepend}$1"/>
      </condition>
    </extension>
  </context>
  <context name="context_2">
    <extension name="conditioning_callerid" continue="true">
      <condition field="${internal_caller_id_number}" expression="^.+$">
        <action application="set" data="effective_caller_id_name=${internal_caller_id_name}"/>
        <action application="set" data="effective_caller_id_number=${internal_caller_id_number}"/>
      </condition>
    </extension>
    <extension name="postroute_global" continue="true">
      <condition>
        <action application="hash" data="insert/${domain_name}-spymap/${caller_id_number}/${uuid}"/>
        <action application="hash" data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/>
        <action application="hash" data="insert/${domain_name}-last_dial/global/${uuid}"/>
        <action application="set" data="RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}"/>
      </condition>
    </extension>
    <extension name="main_number_3" continue="true">
      <condition field="destination_number" expression="^502$">
        <action application="set" bluebox="settingEndBridge" data="hangup_after_bridge=true"/>
        <action application="set" bluebox="settingFail" data="continue_on_fail=true"/>
        <action application="set" bluebox="autoPlay" data="vm_auto_play=false"/>
        <action application="set" bluebox="settingTimeout" data="call_timeout=30"/>
        <action application="set" bluebox="settingRing" data="ringback=${us-ring}"/>
        <action application="set" bluebox="settingRingXfer" data="transfer_ringback=${us-ring}"/>
        <action application="export" bluebox="sipCalleeIdName" data="sip_callee_id_name=Monkey"/>
        <action application="export" bluebox="sipCalleeIdNumber" data="sip_callee_id_number=502"/>
        <action application="bridge" data="user/502@$${location_1}"/>
        <action application="answer"/>
        <action application="voicemail" data="default voicemail_1 2001"/>
        <action application="hangup"/>
      </condition>
    </extension>
    <extension name="main_number_4" continue="true">
      <condition field="destination_number" expression="^17036526678$">
        <action application="set" bluebox="settingTimeout" data="call_timeout=30"/>
        <action application="set" bluebox="settingRing" data="ringback=${us-ring}"/>
        <action application="set" bluebox="settingRingXfer" data="transfer_ringback=${us-ring}"/>
        <action application="export" bluebox="sipCalleeIdName" data="sip_callee_id_name=Account Admin"/>
        <action application="export" bluebox="sipCalleeIdNumber" data="sip_callee_id_number=17036526678"/>
        <action application="bridge" data="user/account_admin@$${location_1}"/>
        <action application="hangup"/>
      </condition>
    </extension>
    <extension name="main_trunk_1_pattern_1" continue="true">
      <condition field="destination_number" break="never" expression="^1{0,1}([2-9][0-8][0-9][2-9][0-9]{6})$" bluebox="pattern_1_part_1">
        <action application="set" bluebox="prepend" data="prepend="/>
        <action application="set" bluebox="cid_name" data="effective_caller_id_name=Statira Systems"/>
        <action application="set" bluebox="cid_number" data="effective_caller_id_number=7036526678"/>
      </condition>
      <condition field="${outbound_caller_id_number}" expression="^.+$" break="never" bluebox="caller_id">
        <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>
        <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
        <action application="export" bluebox="caller_id_field" data="sip_cid_type=rpid"/>
      </condition>
      <condition field="destination_number" break="never" expression="^1{0,1}([2-9][0-8][0-9][2-9][0-9]{6})$" bluebox="pattern_1_part_1_out">
        <action application="bridge" bluebox="out_trunk_1" data="sofia/gateway/trunk_1/${prepend}$1"/>
      </condition>
    </extension>
  </context>
</include>

/opt/freeswitch/conf/dialplan/bluebox_routes.xml
<?xml version="1.0"?>
<include>
  <context name="multitenant_routing_context">
    <extension name="route_number_4" continue="true">
      <condition field="destination_number" expression="^17036526678$">
        <action application="set" data="vm-operator-extension=17036526678"/>
        <action application="set" data="force_transfer_context=context_2"/>
        <action application="transfer" data="17036526678 XML context_2"/>
      </condition>
    </extension>
  </context>
</include>

William J. Moore
Owner
Statira Systems
611 Caroline St
Fredericksburg, VA 22401
540.693.0579
www.statirasystems.com<http://www.statirasystems.com>

On 6/22/2011 12:34 PM, William Moore wrote:
I can't get any incoming nor outgoing calls. The trunk is through Junction Networks and registers fine. I though I had the dial plan correct but there is obviously something I am missing.

I am running Blue.Box GUI with FreeSwitch. It is in CentOS 5.5. I am using Verizon FiOS internet and the firewall is properly configured. This is a test environment so I would be happy to let someone in to take a look. I have SSH setup and web access via phone.statirasystems.com. Other wise it is a lot of configuration info to post.

I have checked with Junction Networks and everything is fine on there end. My server is not responding to the requests.

I am fairly green with freeswitch and would appreciate any help.

Please note: I have sought assistance via the 2600hz community to no avail.

Thank you,



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

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

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From rhuddleston at gmail.com  Fri Jun 24 05:53:37 2011
From: rhuddleston at gmail.com (Robert-iPhone)
Date: Thu, 23 Jun 2011 21:53:37 -0400
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <CA293E4C.18131%krice@freeswitch.org>
References: <CA293E4C.18131%krice@freeswitch.org>
Message-ID: <32B9468B-3402-4056-A18D-D4A01626A87A@gmail.com>

isnt voip great - nobody learns telecom 101.
LERG
LRN
LNP
OCN
B8ZS / SF

just the good ole days 5ESS / DMS RCV

Sent from my iPhone

On Jun 23, 2011, at 8:02 PM, Ken Rice <krice at freeswitch.org> wrote:

> That?s called the LRN database and its not free... There are services that will offer you access to it on a per query basis out there tho
> 
> K
> 
> 
> On 6/23/11 6:59 PM, "Chris Cureau" <cmcureau at gmail.com> wrote:
> 
> I'm guessing this is just a pipe dream, but I'll throw it out anyway...
> 
> Is there any database available that can determine the carrier for an outbound number? It would be really nice to be able to route cellular calls via a cell phone automatically instead of setting up specific numbers in the profile.
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From david.ponzone at ipeva.fr  Fri Jun 24 05:56:04 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 24 Jun 2011 03:56:04 +0200
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <CA2932CA.2B657%d@d-man.org>
References: <CA2932CA.2B657%d@d-man.org>
Message-ID: <73317B9D-B8B3-4F1B-AD2A-7F9B3DF1DDDC@ipeva.fr>

No Darren, sorry, I confused FreePBX and FusionPBX.

My bad.

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

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

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




Le 24/06/2011 ? 03:14, Darren Schreiber a ?crit :

> David,
> 	FreePBX v3 is now called blue.box. You're not making any sense? Is this an attempt at creating FUD?
> 
> - Darren
> 
> -- 
> 
> 
> From: David Ponzone <david.ponzone at ipeva.fr>
> Reply-To: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>
> Date: Thu, 23 Jun 2011 12:59:16 -0700
> To: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Trunk Issue
> 
> I had the same feeling about blue.box.
> I think there are more interesting GUIs out there, like FreePBX V3.
> The weird thing is that none of them seems very active.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 23/06/2011 ? 19:55, William Moore a ?crit :
> 
>> I give up on blue.box. They seem preoccupied by there other projects. Documentation does not explain enough. It is like describing a human by their parts only and not how it interacts. Peace out.
>> William J. Moore
>> Statira Systems
>> 611 Caroline St
>> Fredericksburg, VA
>> http://www.statirasystems.com
>> 
>> On 06/23/2011 01:34 PM, David Ponzone wrote:
>>> 
>>> In sipinterface_2, make the following modification:
>>> 
>>>  <param name="auth-calls" value="false"/> 
>>> 
>>> 
>>>         
>>> David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>> 
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>> 
>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>> 
>>> 
>>> 
>>> 
>>> Le 23/06/2011 ? 19:31, David Ponzone a ?crit :
>>> 
>>>> The issue is that you are expecting them to authenticate, but they won't.
>>>> 
>>>> David Ponzone  Direction Technique
>>>> email: david.ponzone at ipeva.fr
>>>> tel:      01 74 03 18 97
>>>> gsm:   06 66 98 76 34
>>>> 
>>>> Service Client IPeva
>>>> tel:      0811 46 26 26
>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>> 
>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Le 23/06/2011 ? 19:15, William Moore a ?crit :
>>>> 
>>>>> Ok, I reregistered and called right away. I get the following:
>>>>>   0.000000 66.227.100.20 -> 192.168.1.254 SIP/SDP Request: INVITE sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with session description
>>>>>   0.000766 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>>>>>   0.002321 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy Authentication Required
>>>>>   0.021120 66.227.100.20 -> 192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>>>>   3.272168 192.168.1.254 -> 66.227.100.20 SIP Request: REGISTER sip:jnctn.net;transport=udp
>>>>>   3.292076 66.227.100.20 -> 192.168.1.254 SIP Status: 200 OK    (1 bindings)
>>>>> 
>>>>> The call just states it is a none working number. That means something in my dial plan is incorrect. I am double checking it now.
>>>>> William J. MooreStatira Systems
>>>>> 611 Caroline St
>>>>> Fredericksburg, VA
>>>>> http://www.statirasystems.com
>>>>> 
>>>>> On 06/23/2011 12:58 PM, William Moore wrote:
>>>>>> 
>>>>>> When I put it in and it reloads, blue.box removes it. Can I reference the gateway in another sip interface file or does it have to be in that one?
>>>>>> William J. Moore
>>>>>> Statira Systems
>>>>>> 611 Caroline St
>>>>>> Fredericksburg, VA
>>>>>> http://www.statirasystems.com
>>>>>> 
>>>>>> On 06/23/2011 12:52 PM, William Moore wrote:
>>>>>>> 
>>>>>>> correction: missed a line.
>>>>>>> 
>>>>>>>   0.000000 66.227.100.20 -> 192.168.1.254 SIP/SDP Request: INVITE sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with session description
>>>>>>>   0.000814 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>>>>>>>   0.002440 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy Authentication Required
>>>>>>>   0.023173 66.227.100.20 -> 192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>>>>>> 
>>>>>>> William J. Moore
>>>>>>> Statira Systems
>>>>>>> 611 Caroline St
>>>>>>> Fredericksburg, VA
>>>>>>> http://www.statirasystems.com
>>>>>>> 
>>>>>>> On 06/23/2011 12:50 PM, William Moore wrote:
>>>>>>>> 
>>>>>>>> I get this with wireshark
>>>>>>>> 
>>>>>>>>  40.345679 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>>>>>>>>  40.347381 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy Authentication Required
>>>>>>>>  40.367906 66.227.100.20 -> 192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>>>>>>> 
>>>>>>>> I am looking for the expiration now..
>>>>>>>> William J. Moore
>>>>>>>> Statira Systems
>>>>>>>> 611 Caroline St
>>>>>>>> Fredericksburg, VA
>>>>>>>> http://www.statirasystems.com
>>>>>>>> 
>>>>>>>> On 06/23/2011 12:25 PM, David Ponzone wrote:
>>>>>>>>> 
>>>>>>>>> Take a sip trace on your box with tcpdump or preferably, tshark:
>>>>>>>>> tshark port 5080
>>>>>>>>> 
>>>>>>>>> You will then see if you receive it.
>>>>>>>>> 
>>>>>>>>> In the config you sent, I don't see anything about register expiration, or NAT keepalive, so I would really recommend you add a ping every 30 seconds to your gateway config.
>>>>>>>>> 
>>>>>>>>> If the issue is there, it's quite easy to see:
>>>>>>>>> unregister the gateway
>>>>>>>>> register it again
>>>>>>>>> make an inbound call right away
>>>>>>>>> 
>>>>>>>>> David Ponzone  Direction Technique
>>>>>>>>> email: david.ponzone at ipeva.fr
>>>>>>>>> tel:      01 74 03 18 97
>>>>>>>>> gsm:   06 66 98 76 34
>>>>>>>>> 
>>>>>>>>> Service Client IPeva
>>>>>>>>> tel:      0811 46 26 26
>>>>>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>>>>>> 
>>>>>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> Le 23/06/2011 ? 18:15, William Moore a ?crit :
>>>>>>>>> 
>>>>>>>>>> It was the 1 and it needs to be set for e.164(+1). Blue.box makes it a little more difficult since it wants to manage the xmls. I could have set it up as a separate config file however that defeats the purpose of the GUI.
>>>>>>>>>> 
>>>>>>>>>> I still have the issue of incoming calls not even showing up in sofia. Junction Networks says it is going out, but my server is not responding. I'm not sure if I have the settings for incoming correct.
>>>>>>>>>> William J. Moore
>>>>>>>>>> Statira Systems
>>>>>>>>>> 611 Caroline St
>>>>>>>>>> Fredericksburg, VA
>>>>>>>>>> http://www.statirasystems.com
>>>>>>>>>> 
>>>>>>>>>> On 06/23/2011 04:21 AM, David Ponzone wrote:
>>>>>>>>>>> 
>>>>>>>>>>> William,
>>>>>>>>>>> 
>>>>>>>>>>> If I believe the example configuraiton on Junction's web, they don't expect E164:
>>>>>>>>>>> 
>>>>>>>>>>>     <extension name="junctionnetworks.com>
>>>>>>>>>>>     <condition field="destination_number" expression="^(1{0,1}\d{10})$">
>>>>>>>>>>>       <action application="set"data="effective_caller_id_number=OUTBOUNDCALLERID"/>
>>>>>>>>>>>       <action application="bridge" data="sofia/gateway/junctionnetworks.com/$1"/>
>>>>>>>>>>>      </condition>
>>>>>>>>>>>    </extension>
>>>>>>>>>>> But, elsewhere in their knowledgebase, they say:
>>>>>>>>>>> 
>>>>>>>>>>> Numbering
>>>>>>>>>>> The standard e.164 numbering plan (ITU) is used. North American numbers are required to be prefixed with a '1'. International numbers need to be prefixed with "011".
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>> I think they are confused about what is E164....
>>>>>>>>>>> 
>>>>>>>>>>> You should try to prefix with 1 as Michael recommended.
>>>>>>>>>>> 
>>>>>>>>>>> David Ponzone  Direction Technique
>>>>>>>>>>> email: david.ponzone at ipeva.fr
>>>>>>>>>>> tel:      01 74 03 18 97
>>>>>>>>>>> gsm:   06 66 98 76 34
>>>>>>>>>>> 
>>>>>>>>>>> Service Client IPeva
>>>>>>>>>>> tel:      0811 46 26 26
>>>>>>>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>>>>>>>> 
>>>>>>>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>> Le 23/06/2011 ? 08:19, Michael Collins a ?crit :
>>>>>>>>>>> 
>>>>>>>>>>>> Perhaps the gateway wants the leading 1 or expects e.164 format?
>>>>>>>>>>>> -MC
>>>>>>>>>>>> 
>>>>>>>>>>>> On Wed, Jun 22, 2011 at 5:37 PM, Brian West <brian at freeswitch.org> wrote:
>>>>>>>>>>>>> 484 is address incomplete means you didn't send enough digits to the gateway usually.
>>>>>>>>>>>>> 
>>>>>>>>>>>>> /b
>>>>>>>>>>>>> 
>>>>>>>>>>>>> On Jun 22, 2011, at 7:34 PM, Brad Mina wrote:
>>>>>>>>>>>>> 
>>>>>>>>>>>>>> Ah, thought it was recv not send. Good to know that an invalid gateway issue
>>>>>>>>>>>>>> can cause this.
>>>>>>>>>>>>>> 
>>>>>>>>>>>>>> On Wed, Jun 22, 2011 at 5:31 PM, Michael Collins <msc at freeswitch.org> wrote:
>>>>>>>>>>>>>> 
>>>>>>>>>>>>>>> The 484 I believe is a result of the bad gateway. In any case, the OP needs
>>>>>>>>>>>>>>> to figure out what's up w/ that gw or no calls will be going out...
>>>>>>>>>>>>>>> 
>>>>>>>>>>>>>>> -MC
>>>>>>>>>>>>>> 
>>>>>>>>>>>>> 
>>>>>>>>>>>>> 
>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>>> 
>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>> 
>>>>>>>>>>>> 
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>> 
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>> 
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>> 
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>> 
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>>>> 
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>> 
>>> 
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From d at d-man.org  Fri Jun 24 06:05:45 2011
From: d at d-man.org (Darren Schreiber)
Date: Thu, 23 Jun 2011 19:05:45 -0700
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <73317B9D-B8B3-4F1B-AD2A-7F9B3DF1DDDC@ipeva.fr>
Message-ID: <CA293ED7.2B6A6%d@d-man.org>

OK, well again, I'm happy to help, as is my team and as are the open-source folks. Sometimes we do get busy, so please just ping me, but the logs pasted by William indicate he doesn't know how to setup the authentication for his carriers/phones and since there's 82 different ways to do that, I'd need to know more info.

Hence the request to email off list, unless he wants to post his auth strategy and IPs publicly :)

--


From: David Ponzone <david.ponzone at ipeva.fr<mailto:david.ponzone at ipeva.fr>>
Reply-To: "freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>" <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Date: Thu, 23 Jun 2011 18:56:04 -0700
To: "freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>" <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] Trunk Issue

No Darren, sorry, I confused FreePBX and FusionPBX.

My bad.

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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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




Le 24/06/2011 ? 03:14, Darren Schreiber a ?crit :

David,
FreePBX v3 is now called blue.box. You're not making any sense? Is this an attempt at creating FUD?

- Darren

--


From: David Ponzone <david.ponzone at ipeva.fr<mailto:david.ponzone at ipeva.fr>>
Reply-To: "freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>" <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Date: Thu, 23 Jun 2011 12:59:16 -0700
To: "freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>" <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] Trunk Issue

I had the same feeling about blue.box.
I think there are more interesting GUIs out there, like FreePBX V3.
The weird thing is that none of them seems very active.

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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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




Le 23/06/2011 ? 19:55, William Moore a ?crit :

I give up on blue.box. They seem preoccupied by there other projects. Documentation does not explain enough. It is like describing a human by their parts only and not how it interacts. Peace out.

William J. Moore
Statira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com<http://www.statirasystems.com/>

On 06/23/2011 01:34 PM, David Ponzone wrote:
In sipinterface_2, make the following modification:

 <param name="auth-calls" value="false"/>

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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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




Le 23/06/2011 ? 19:31, David Ponzone a ?crit :

The issue is that you are expecting them to authenticate, but they won't.

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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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




Le 23/06/2011 ? 19:15, William Moore a ?crit :

Ok, I reregistered and called right away. I get the following:

  0.000000 66.227.100.20 -> 192.168.1.254 SIP/SDP Request: INVITE sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with session description
  0.000766 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
  0.002321 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy Authentication Required
  0.021120 66.227.100.20 -> 192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
  3.272168 192.168.1.254 -> 66.227.100.20 SIP Request: REGISTER sip:jnctn.net;transport=udp
  3.292076 66.227.100.20 -> 192.168.1.254 SIP Status: 200 OK    (1 bindings)

The call just states it is a none working number. That means something in my dial plan is incorrect. I am double checking it now.


William J. MooreStatira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com<http://www.statirasystems.com/>

On 06/23/2011 12:58 PM, William Moore wrote:
When I put it in and it reloads, blue.box removes it. Can I reference the gateway in another sip interface file or does it have to be in that one?

William J. Moore
Statira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com<http://www.statirasystems.com/>

On 06/23/2011 12:52 PM, William Moore wrote:
correction: missed a line.

  0.000000 66.227.100.20 -> 192.168.1.254 SIP/SDP Request: INVITE sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with session description
  0.000814 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
  0.002440 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy Authentication Required
  0.023173 66.227.100.20 -> 192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2


William J. Moore
Statira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com<http://www.statirasystems.com/>

On 06/23/2011 12:50 PM, William Moore wrote:
I get this with wireshark

 40.345679 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
 40.347381 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy Authentication Required
 40.367906 66.227.100.20 -> 192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2

I am looking for the expiration now..

William J. Moore
Statira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com<http://www.statirasystems.com/>

On 06/23/2011 12:25 PM, David Ponzone wrote:
Take a sip trace on your box with tcpdump or preferably, tshark:
tshark port 5080

You will then see if you receive it.

In the config you sent, I don't see anything about register expiration, or NAT keepalive, so I would really recommend you add a ping every 30 seconds to your gateway config.

If the issue is there, it's quite easy to see:
unregister the gateway
register it again
make an inbound call right away

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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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




Le 23/06/2011 ? 18:15, William Moore a ?crit :

It was the 1 and it needs to be set for e.164(+1). Blue.box makes it a little more difficult since it wants to manage the xmls. I could have set it up as a separate config file however that defeats the purpose of the GUI.

I still have the issue of incoming calls not even showing up in sofia. Junction Networks says it is going out, but my server is not responding. I'm not sure if I have the settings for incoming correct.

William J. Moore
Statira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com<http://www.statirasystems.com/>

On 06/23/2011 04:21 AM, David Ponzone wrote:
William,

If I believe the example configuraiton on Junction's web, they don't expect E164:


   <extension name="junctionnetworks.com<http://junctionnetworks.com/>>
    <condition field="destination_number" expression="^(1{0,1}\d{10})$">
      <action application="set"data="effective_caller_id_number=OUTBOUNDCALLERID"/>
      <action application="bridge" data="sofia/gateway/junctionnetworks.com/$1"/>
     </condition>
   </extension>


But, elsewhere in their knowledgebase, they say:


Numbering
The standard e.164 numbering plan (ITU<http://www.itu.int/>) is used. North American numbers are required to be prefixed with a '1'. International numbers need to be prefixed with "011".

I think they are confused about what is E164....

You should try to prefix with 1 as Michael recommended.

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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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




Le 23/06/2011 ? 08:19, Michael Collins a ?crit :

Perhaps the gateway wants the leading 1 or expects e.164 format?
-MC

On Wed, Jun 22, 2011 at 5:37 PM, Brian West <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:
484 is address incomplete means you didn't send enough digits to the gateway usually.

/b

On Jun 22, 2011, at 7:34 PM, Brad Mina wrote:

Ah, thought it was recv not send. Good to know that an invalid gateway issue
can cause this.

On Wed, Jun 22, 2011 at 5:31 PM, Michael Collins <msc at freeswitch.org<mailto:msc at freeswitch.org>> wrote:

The 484 I believe is a result of the bad gateway. In any case, the OP needs
to figure out what's up w/ that gw or no calls will be going out...

-MC



_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
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_______________________________________________
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From cmcureau at gmail.com  Fri Jun 24 06:12:15 2011
From: cmcureau at gmail.com (Chris Cureau)
Date: Thu, 23 Jun 2011 21:12:15 -0500
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <32B9468B-3402-4056-A18D-D4A01626A87A@gmail.com>
References: <CA293E4C.18131%krice@freeswitch.org>
	<32B9468B-3402-4056-A18D-D4A01626A87A@gmail.com>
Message-ID: <1769702213133480228@unknownmsgid>

Consider this my telephony 101 experience. The past 15 years have been
filled with tech support, coding, and project management. I guess I was
looking for a change.

If one can't ask even newbie questions without this type of disdain, how
does one get the knowledge?

On Jun 23, 2011, at 8:54 PM, Robert-iPhone <rhuddleston at gmail.com> wrote:

isnt voip great - nobody learns telecom 101.
LERG
LRN
LNP
OCN
B8ZS / SF

just the good ole days 5ESS / DMS RCV

Sent from my iPhone

On Jun 23, 2011, at 8:02 PM, Ken Rice <krice at freeswitch.org> wrote:

That?s called the LRN database and its not free... There are services that
will offer you access to it on a per query basis out there tho

K


On 6/23/11 6:59 PM, "Chris Cureau" < <cmcureau at gmail.com>cmcureau at gmail.com>
wrote:

I'm guessing this is just a pipe dream, but I'll throw it out anyway...

Is there any database available that can determine the carrier for an
outbound number? It would be really nice to be able to route cellular calls
via a cell phone automatically instead of setting up specific numbers in the
profile.

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

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 <FreeSWITCH-users at lists.freeswitch.org>
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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
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From rhuddleston at gmail.com  Fri Jun 24 06:27:34 2011
From: rhuddleston at gmail.com (Robert-iPhone)
Date: Thu, 23 Jun 2011 22:27:34 -0400
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <1769702213133480228@unknownmsgid>
References: <CA293E4C.18131%krice@freeswitch.org>
	<32B9468B-3402-4056-A18D-D4A01626A87A@gmail.com>
	<1769702213133480228@unknownmsgid>
Message-ID: <3DF51B5A-A97F-42AE-9C31-90D132B4F066@gmail.com>

it wasnt intended as insult - just humorous how voip requires less knowledge

Sent from my iPhone

On Jun 23, 2011, at 10:12 PM, Chris Cureau <cmcureau at gmail.com> wrote:

> Consider this my telephony 101 experience. The past 15 years have been filled with tech support, coding, and project management. I guess I was looking for a change.
> 
> If one can't ask even newbie questions without this type of disdain, how does one get the knowledge?
> 
> On Jun 23, 2011, at 8:54 PM, Robert-iPhone <rhuddleston at gmail.com> wrote:
> 
>> isnt voip great - nobody learns telecom 101.
>> LERG
>> LRN
>> LNP
>> OCN
>> B8ZS / SF
>> 
>> just the good ole days 5ESS / DMS RCV
>> 
>> Sent from my iPhone
>> 
>> On Jun 23, 2011, at 8:02 PM, Ken Rice <krice at freeswitch.org> wrote:
>> 
>>> That?s called the LRN database and its not free... There are services that will offer you access to it on a per query basis out there tho
>>> 
>>> K
>>> 
>>> 
>>> On 6/23/11 6:59 PM, "Chris Cureau" <cmcureau at gmail.com> wrote:
>>> 
>>> I'm guessing this is just a pipe dream, but I'll throw it out anyway...
>>> 
>>> Is there any database available that can determine the carrier for an outbound number? It would be really nice to be able to route cellular calls via a cell phone automatically instead of setting up specific numbers in the profile.
>>> 
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From cmcureau at gmail.com  Fri Jun 24 06:30:10 2011
From: cmcureau at gmail.com (Chris Cureau)
Date: Thu, 23 Jun 2011 21:30:10 -0500
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <3DF51B5A-A97F-42AE-9C31-90D132B4F066@gmail.com>
References: <CA293E4C.18131%krice@freeswitch.org>
	<32B9468B-3402-4056-A18D-D4A01626A87A@gmail.com>
	<1769702213133480228@unknownmsgid>
	<3DF51B5A-A97F-42AE-9C31-90D132B4F066@gmail.com>
Message-ID: <BANLkTinO_GEzHyisWD-ix7i8uK7WX7Bn6g@mail.gmail.com>

My apologies, then :)

On Thu, Jun 23, 2011 at 9:27 PM, Robert-iPhone <rhuddleston at gmail.com>wrote:

> it wasnt intended as insult - just humorous how voip requires less
> knowledge
>
> Sent from my iPhone
>
>
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From frankie.k.yiu at gmail.com  Fri Jun 24 06:35:18 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Thu, 23 Jun 2011 19:35:18 -0700
Subject: [Freeswitch-users] Any RECORD_STOP event sample available?
Message-ID: <BANLkTimkbxVe3GxFfaHY-+LxBs5_umpGSA@mail.gmail.com>

Hi there,

Could someone please point me where I can find a sample of the RECORD_STOP
event look like?
I look at this page but it does not have any information about it. (
http://wiki.freeswitch.org/wiki/Event_list#RECORD_STOP)

Thanks,
Frankie
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From max.asterisk at gmail.com  Fri Jun 24 08:32:56 2011
From: max.asterisk at gmail.com (Max Alex)
Date: Fri, 24 Jun 2011 10:02:56 +0530
Subject: [Freeswitch-users] Answer issue on inbound call
In-Reply-To: <BANLkTi=Q3qRBCDRARDWp15=Guwyq-jtNLQ@mail.gmail.com>
References: <BANLkTin+f=uDMiOouyLWAorPvNTd1Z6xaA@mail.gmail.com>
	<BANLkTim=TOQD1tUvmHjeRuhAR-eKkGYgsw@mail.gmail.com>
	<BANLkTim3pTPV804t_Usk59eG2C56HAyFdA@mail.gmail.com>
	<BANLkTi=Q3qRBCDRARDWp15=Guwyq-jtNLQ@mail.gmail.com>
Message-ID: <BANLkTinMokGeJU+wVVA0U-oxB97z+c+ZGg@mail.gmail.com>

Hi,
Thanks for reply.
Current scenario is below.

PSTN call -> sangoma device -> freeswitch incoming context -> routed to 1001
-> ringing (Answered on cellphone)
Here when it is routed to 1001 the call it is started ringing, but on phone
that call is answered and hearding sound of ringing.

Required flow:
PSTN call -> sangoma device -> freeswitch incoming context -> routed to 1001
-> ringing (Ringing on cellphone)

I hope it is understable, the call should not be answered until 1001 answer
it, right not when it is started ring it is answered on cell phone.
that should not be happen as it is not answered yet.

Waiting for your reply.

Thanks,
Max Alex
Voip Developer



On Fri, Jun 24, 2011 at 5:48 AM, Michael Collins <msc at freeswitch.org> wrote:

> I'm not sure I understand the problem. What is happening vs. what you
> believe should be happening?
> -MC
>
>
> On Thu, Jun 23, 2011 at 3:31 AM, Max Alex <max.asterisk at gmail.com> wrote:
>
>> Hi,
>> Thanks for your reply.
>> Here is my configuration and log
>> http://pastebin.freeswitch.org/16571
>>
>> I am using A200 analog sangoma device with freeswitch, it is working fine
>> but when it is routing call to 1001 then it is answered.
>> Please provider your suggestions.
>>
>> Thanks,
>> Max Alex
>> Voip Developer
>>
>>
>>
>>
>> On Wed, Jun 22, 2011 at 8:51 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> I thought the A200 was an analog card? Maybe I have my numbers mixed
>>> up...
>>>
>>> Go ahead and collect a debug log of this call. It might help to have your
>>> configs posted as well. Use pastebin.freeswitch.org. See this wiki
>>> article for tips on how to collect information:
>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>>
>>> -MC
>>>
>>> On Wed, Jun 22, 2011 at 3:23 AM, Max Alex <max.asterisk at gmail.com>wrote:
>>>
>>>> Hi,
>>>> I have installed freeswitch latest version with sangoma card A200 as
>>>> well,
>>>> I have installed and configured freetdm module with wanpipe drivers for
>>>> freeswitch,
>>>> We are properly receiving the inbound calls in public context and then
>>>> we are routing that call to 1001 extension,
>>>> it is properly routing to 1001 as well, but we have one issue while
>>>> routing on 1001.
>>>>
>>>> Here is the issue description.
>>>> I am calling from my cell phone to that DID number of pri line, and then
>>>> it will start ringing on 1001 extension,
>>>> When 1001 extension start ringing the call is answered on my cell phone,
>>>> it is something like freeswitch preanswer or autoanswer the call, how
>>>> can i stop this answer call when it is ringing on 1001 extension,
>>>> Waiting for good reply.
>>>>
>>>> Thanks,
>>>> Max Alex
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
>
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From msc at freeswitch.org  Fri Jun 24 10:08:45 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 23 Jun 2011 23:08:45 -0700
Subject: [Freeswitch-users] Answer issue on inbound call
In-Reply-To: <BANLkTinMokGeJU+wVVA0U-oxB97z+c+ZGg@mail.gmail.com>
References: <BANLkTin+f=uDMiOouyLWAorPvNTd1Z6xaA@mail.gmail.com>
	<BANLkTim=TOQD1tUvmHjeRuhAR-eKkGYgsw@mail.gmail.com>
	<BANLkTim3pTPV804t_Usk59eG2C56HAyFdA@mail.gmail.com>
	<BANLkTi=Q3qRBCDRARDWp15=Guwyq-jtNLQ@mail.gmail.com>
	<BANLkTinMokGeJU+wVVA0U-oxB97z+c+ZGg@mail.gmail.com>
Message-ID: <BANLkTinEp1Cu5mB_UHw-Y+mitGsNByR3sQ@mail.gmail.com>

Are you using the default dialplan? I think you might just need to ignore
early media on your bridge app. If you are using the default.xml file then
locate "Local_Extension" and the bridge line:

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

Change it to this, then try again:

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

If I understand correctly, the "symptom" you are experiencing is the normal
operation of the bridge app (and it's cousin, the originate API command).
When the b-leg sends back media, including ringing, the bridge (or the
originate) will be considered "successful," and in the case of bridge, the
b-leg's audio (the early media) will be connected to the a-leg. If you set
ignore_early_media=true then you are explicitly telling the bridge app that
you only want to connect the b-leg to the a-leg if the b-leg actually
answers.

I hope that made sense...

-MC


On Thu, Jun 23, 2011 at 9:32 PM, Max Alex <max.asterisk at gmail.com> wrote:

> Hi,
> Thanks for reply.
> Current scenario is below.
>
> PSTN call -> sangoma device -> freeswitch incoming context -> routed to
> 1001 -> ringing (Answered on cellphone)
> Here when it is routed to 1001 the call it is started ringing, but on phone
> that call is answered and hearding sound of ringing.
>
> Required flow:
> PSTN call -> sangoma device -> freeswitch incoming context -> routed to
> 1001 -> ringing (Ringing on cellphone)
>
> I hope it is understable, the call should not be answered until 1001 answer
> it, right not when it is started ring it is answered on cell phone.
> that should not be happen as it is not answered yet.
>
> Waiting for your reply.
>
>
> Thanks,
> Max Alex
> Voip Developer
>
>
>
> On Fri, Jun 24, 2011 at 5:48 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> I'm not sure I understand the problem. What is happening vs. what you
>> believe should be happening?
>> -MC
>>
>>
>> On Thu, Jun 23, 2011 at 3:31 AM, Max Alex <max.asterisk at gmail.com> wrote:
>>
>>> Hi,
>>> Thanks for your reply.
>>> Here is my configuration and log
>>> http://pastebin.freeswitch.org/16571
>>>
>>> I am using A200 analog sangoma device with freeswitch, it is working fine
>>> but when it is routing call to 1001 then it is answered.
>>> Please provider your suggestions.
>>>
>>> Thanks,
>>> Max Alex
>>> Voip Developer
>>>
>>>
>>>
>>>
>>> On Wed, Jun 22, 2011 at 8:51 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> I thought the A200 was an analog card? Maybe I have my numbers mixed
>>>> up...
>>>>
>>>> Go ahead and collect a debug log of this call. It might help to have
>>>> your configs posted as well. Use pastebin.freeswitch.org. See this wiki
>>>> article for tips on how to collect information:
>>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>>>
>>>> -MC
>>>>
>>>> On Wed, Jun 22, 2011 at 3:23 AM, Max Alex <max.asterisk at gmail.com>wrote:
>>>>
>>>>> Hi,
>>>>> I have installed freeswitch latest version with sangoma card A200 as
>>>>> well,
>>>>> I have installed and configured freetdm module with wanpipe drivers for
>>>>> freeswitch,
>>>>> We are properly receiving the inbound calls in public context and then
>>>>> we are routing that call to 1001 extension,
>>>>> it is properly routing to 1001 as well, but we have one issue while
>>>>> routing on 1001.
>>>>>
>>>>> Here is the issue description.
>>>>> I am calling from my cell phone to that DID number of pri line, and
>>>>> then it will start ringing on 1001 extension,
>>>>> When 1001 extension start ringing the call is answered on my cell
>>>>> phone,
>>>>> it is something like freeswitch preanswer or autoanswer the call, how
>>>>> can i stop this answer call when it is ringing on 1001 extension,
>>>>> Waiting for good reply.
>>>>>
>>>>> Thanks,
>>>>> Max Alex
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
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>
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From msc at freeswitch.org  Fri Jun 24 10:11:31 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 23 Jun 2011 23:11:31 -0700
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <BANLkTinO_GEzHyisWD-ix7i8uK7WX7Bn6g@mail.gmail.com>
References: <CA293E4C.18131%krice@freeswitch.org>
	<32B9468B-3402-4056-A18D-D4A01626A87A@gmail.com>
	<1769702213133480228@unknownmsgid>
	<3DF51B5A-A97F-42AE-9C31-90D132B4F066@gmail.com>
	<BANLkTinO_GEzHyisWD-ix7i8uK7WX7Bn6g@mail.gmail.com>
Message-ID: <BANLkTinbNG6nF5+YGfFLB3T8uf4G7ckcGQ@mail.gmail.com>

On Thu, Jun 23, 2011 at 7:30 PM, Chris Cureau <cmcureau at gmail.com> wrote:

> My apologies, then :)
>
> No worries. Besides, you didn't miss anything interesting. :D
-MC
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From mays.david at gmail.com  Fri Jun 24 10:26:44 2011
From: mays.david at gmail.com (David Ma)
Date: Fri, 24 Jun 2011 14:26:44 +0800
Subject: [Freeswitch-users] Leg-A is automatically disconnected on Leg-B
 orginate failure
In-Reply-To: <BANLkTikhPmjN5HeeJHBnQt1ivA8zbtCnYA@mail.gmail.com>
References: <1308213151543-6482192.post@n2.nabble.com>
	<BANLkTi=xYgBTPVGxVGmk9f2=bmX5X2vF=Q@mail.gmail.com>
	<BANLkTikhPmjN5HeeJHBnQt1ivA8zbtCnYA@mail.gmail.com>
Message-ID: <BANLkTi=m7oMix1ZRC9URtRuAbsG2fWyhMQ@mail.gmail.com>

Hi Michael,

Unfortunately this problem still happens.

I enabled "continue_on_fail" for leg-A when I originated the call. Leg-A
call went well. Then I originated leg-B call ("continue_on_fail" is NOT set
for leg-B), which failed for [DESTINATION_OUT_OF_ORDER]. As the consequence,
leg-A was hung up by FS automatically for [ORIGINATOR_CANCEL].

The log excerpt follows.

Do you think "continue_on_fail" should be also enabled for leg-B call?

Thanks,
D.Ma

2011-06-24 13:29:35.887830 [DEBUG] switch_ivr_originate.c:1971 variable
string 5 = [continue_on_fail=true]

2011-06-24 13:29:35.887830 [NOTICE] switch_channel.c:808 New Channel
sofia/external/03996597632298 at 203.208.207.212[c0bd700d-913c-42ad-b68f-81001bf658b8]
[...]
2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:372
(sofia/external/03996563750911 at 203.208.207.212) State SOFT_EXECUTE going to
sleep
2011-06-24 13:29:48.161779 [DEBUG] switch_channel.c:2535 (sofia/external/
03996563750911 at 203.208.207.212) Callstate Change EARLY -> HANGUP
2011-06-24 13:29:48.161779 [NOTICE] switch_ivr_originate.c:1045 Hangup
sofia/external/03996563750911 at 203.208.207.212 [CS_SOFT_EXECUTE]
[DESTINATION_OUT_OF_ORDER]
2011-06-24 13:29:48.161779 [DEBUG] switch_channel.c:2551 Send signal
sofia/external/03996563750911 at 203.208.207.212 [KILL]
2011-06-24 13:29:48.161779 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996563750911 at 203.208.207.212 [BREAK]
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996563750911 at 203.208.207.212) Running State Change
CS_HANGUP
2011-06-24 13:29:48.161779 [DEBUG] switch_channel.c:2535 (sofia/external/
03996597632298 at 203.208.207.212) Callstate Change ACTIVE -> HANGUP
2011-06-24 13:29:48.161779 [NOTICE] switch_ivr_bridge.c:772 Hangup
sofia/external/03996597632298 at 203.208.207.212 [CS_SOFT_EXECUTE]
[ORIGINATOR_CANCEL]
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:557
(sofia/external/03996563750911 at 203.208.207.212) State HANGUP
2011-06-24 13:29:48.161779 [DEBUG] mod_sofia.c:457 Channel sofia/external/
03996563750911 at 203.208.207.212 hanging up, cause: DESTINATION_OUT_OF_ORDER
2011-06-24 13:29:48.161779 [DEBUG] mod_sofia.c:510 Sending CANCEL to
sofia/external/03996563750911 at 203.208.207.212
2011-06-24 13:29:48.161779 [DEBUG] switch_channel.c:2551 Send signal
sofia/external/03996597632298 at 203.208.207.212 [KILL]
2011-06-24 13:29:48.161779 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996597632298 at 203.208.207.212 [BREAK]

On Fri, Jun 17, 2011 at 10:52 AM, David Ma <mays.david at gmail.com> wrote:

> Hi Michael,
>
> Thanks very much for your prompt response! I appreciate the information
> provided.
>
> I was actually searching the the existence of such a variable. I was not so
> luck to find it out and thereby resort to the support forum.
>
> I've modified my code to build this parameter into my application. Will
> feedback to you after verification.
>
> Thanks again,
> D.Ma
>
> On Fri, Jun 17, 2011 at 4:51 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> How about setting this?
>> http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
>>
>> -MC
>>
>>
>> On Thu, Jun 16, 2011 at 1:32 AM, dma <mays.david at gmail.com> wrote:
>>
>>> I am creating a call-back solution. After leg-A answers, I originate
>>> leg-B
>>> call. After receiving SIP 183 from Leg-B, I bridge the 2 legs. However,
>>> in
>>> some cases, leg-A is automatically disconnected by FreeSwitch on leg-B
>>> failure, for example, DESTINATION_OUT_OF_ORDER. The application is not
>>> given
>>> a chance to handle leg-B failure event. This should not be a correct
>>> scenario because I never set "hangup-after-bridge", which is false by
>>> default.
>>>
>>> The right way should be, FreeSwitch doesn't hang up leg-A automatically,
>>> but
>>> give a chance for the application to decide what to do.
>>>
>>> Please see the logs below:
>>>
>>> =================================================
>>>
>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
>>> string 0 = [origination_caller_id_number=03996563750914]
>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
>>> string 1 = [originate_timeout=30]
>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
>>> string 2 = [ccd_session_id=20110610105829676824]
>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
>>> string 3 = [sip_cid_type=pid]
>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
>>> string 4 = [privacy=yes]
>>> 2011-06-10 11:09:55.370506 [NOTICE] switch_channel.c:808 New Channel
>>> sofia/external/03996590031055 at 203.208.207.212
>>> [ea57b74b-a8c2-4fea-9683-98054dc03a79]
>>> 2011-06-10 11:09:55.370506 [DEBUG] mod_sofia.c:4129
>>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_NEW ->
>>> CS_INIT
>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_core_session.c:1116 Send signal
>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_core_state_machine.c:320
>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>> CS_INIT
>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_core_state_machine.c:356
>>> (sofia/external/03996590031055 at 203.208.207.212) State INIT
>>> 2011-06-10 11:09:55.370506 [DEBUG] mod_sofia.c:84
>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA INIT
>>> send 984 bytes to udp/[203.208.207.212]:5060 at 03:09:55.477997:
>>>
>>> ------------------------------------------------------------------------
>>>   INVITE sip:03996590031055 at 203.208.207.212 SIP/2.0
>>>   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKXjQ7eFpKypy5D
>>>   Max-Forwards: 70
>>>   From: "" &lt;sip:03996563750914 at 202.73.56.46&gt;;tag=H4ZN43UH9S9Qc
>>>   To: &lt;sip:03996590031055 at 203.208.207.212&gt;
>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>   CSeq: 13501633 INVITE
>>>   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
>>>   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>> REGISTER, REFER, NOTIFY
>>>   Supported: timer, precondition, path, replaces
>>>   Allow-Events: talk, hold, refer
>>>   Privacy: none
>>>   Content-Type: application/sdp
>>>   Content-Disposition: session
>>>   Content-Length: 204
>>>   X-FS-Support: update_display
>>>   P-Asserted-Identity: "" &lt;sip:03996563750914 at 202.73.56.46&gt;
>>>
>>>   v=0
>>>   o=FreeSWITCH 1307645395 1307645396 IN IP4 202.73.56.46
>>>   s=FreeSWITCH
>>>   c=IN IP4 202.73.56.46
>>>   t=0 0
>>>   m=audio 30000 RTP/AVP 18 3 101 13
>>>   a=rtpmap:101 telephone-event/8000
>>>   a=fmtp:101 0-16
>>>   a=ptime:20
>>>
>>> ------------------------------------------------------------------------
>>> 2011-06-10 11:09:55.371674 [DEBUG] mod_sofia.c:124
>>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_INIT ->
>>> CS_ROUTING
>>> 2011-06-10 11:09:55.371674 [DEBUG] switch_core_session.c:1116 Send signal
>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>> 2011-06-10 11:09:55.371674 [DEBUG] sofia.c:4646 Channel
>>> sofia/external/03996590031055 at 203.208.207.212 entering state
>>> [calling][0]
>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:356
>>> (sofia/external/03996590031055 at 203.208.207.212) State INIT going to
>>> sleep
>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:320
>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>> CS_ROUTING
>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_channel.c:1657
>>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change DOWN ->
>>> RINGING
>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:359
>>> (sofia/external/03996590031055 at 203.208.207.212) State ROUTING
>>> 2011-06-10 11:09:55.373478 [DEBUG] mod_sofia.c:147
>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA ROUTING
>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_ivr_originate.c:66
>>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_ROUTING
>>> ->
>>> CS_CONSUME_MEDIA
>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_session.c:1116 Send signal
>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:359
>>> (sofia/external/03996590031055 at 203.208.207.212) State ROUTING going to
>>> sleep
>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:320
>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>> CS_CONSUME_MEDIA
>>> 2011-06-10 11:09:55.374485 [DEBUG] switch_core_state_machine.c:378
>>> (sofia/external/03996590031055 at 203.208.207.212) State CONSUME_MEDIA
>>> 2011-06-10 11:09:55.374485 [DEBUG] switch_core_state_machine.c:378
>>> (sofia/external/03996590031055 at 203.208.207.212) State CONSUME_MEDIA
>>> going to
>>> sleep
>>> recv 307 bytes from udp/[203.208.207.212]:5060 at 03:09:55.485130:
>>>
>>> ------------------------------------------------------------------------
>>>   SIP/2.0 100 Trying
>>>   Via: SIP/2.0/UDP
>>> 10.1.1.46:5080;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>   To: &lt;sip:03996590031055 at 203.208.207.212&gt;
>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>   CSeq: 13501633 INVITE
>>>   Content-Length:     0
>>>
>>>
>>> ------------------------------------------------------------------------
>>> recv 669 bytes from udp/[203.208.207.212]:5060 at 03:09:56.677788:
>>>
>>> ------------------------------------------------------------------------
>>>   SIP/2.0 183 Session Progress
>>>   Via: SIP/2.0/UDP
>>> 10.1.1.46:5080;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>   To:
>>> &lt;sip:03996590031055 at 203.208.207.212
>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>   CSeq: 13501633 INVITE
>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
>>> SUBSCRIBE, UPDATE
>>>   Content-Type: application/sdp
>>>   Content-Length:   189
>>>
>>>   v=0
>>>   o=- 421265648 1 IN IP4 203.208.207.219
>>>   s=session
>>>   c=IN IP4 203.208.207.196
>>>   t=0 0
>>>   m=audio 30792 RTP/AVP 18 101
>>>   a=rtpmap:18 G729/8000
>>>   a=rtpmap:101 telephone-event/8000
>>>   a=sendrecv
>>>
>>> ------------------------------------------------------------------------
>>> 2011-06-10 11:09:56.571839 [INFO] sofia.c:729
>>> sofia/external/03996590031055 at 203.208.207.212 Update Callee ID to
>>> "Outbound
>>> Call" <03996590031055>
>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia.c:4646 Channel
>>> sofia/external/03996590031055 at 203.208.207.212 entering state
>>> [proceeding][183]
>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia.c:4657 Remote SDP:
>>> v=0
>>> o=- 421265648 1 IN IP4 203.208.207.219
>>> s=session
>>> c=IN IP4 203.208.207.196
>>> t=0 0
>>> m=audio 30792 RTP/AVP 18 101
>>> a=rtpmap:18 G729/8000
>>> a=rtpmap:101 telephone-event/8000
>>>
>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:4467 Audio Codec Compare
>>> [G729:18:8000:20:8000]/[G729:18:8000:20:8000]
>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:2757 Set Codec
>>> sofia/external/03996590031055 at 203.208.207.212 G729/8000 20 ms 160
>>> samples
>>> 8000 bits
>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:4565 Set 2833 dtmf send
>>> payload to 101
>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:2987 AUDIO RTP
>>> [sofia/external/03996590031055 at 203.208.207.212] 10.1.1.46 port 30000 ->
>>> 203.208.207.196 port 30792 codec: 18 ms: 20
>>> 2011-06-10 11:09:56.571839 [DEBUG] switch_rtp.c:1607 Starting timer
>>> [soft]
>>> 160 bytes per 20ms
>>> 2011-06-10 11:09:56.573731 [DEBUG] sofia_glue.c:3228 Set 2833 dtmf send
>>> payload to 101
>>> 2011-06-10 11:09:56.573731 [DEBUG] sofia_glue.c:3233 Set 2833 dtmf
>>> receive
>>> payload to 101
>>> 2011-06-10 11:09:56.573731 [NOTICE] sofia_glue.c:3680 Pre-Answer
>>> sofia/external/03996590031055 at 203.208.207.212!
>>> 2011-06-10 11:09:56.573731 [DEBUG] switch_channel.c:2627
>>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change RINGING
>>> ->
>>> EARLY
>>> 2011-06-10 11:09:56.573731 [DEBUG] switch_ivr_originate.c:3408 Originate
>>> Resulted in Success: [sofia/external/03996590031055 at 203.208.207.212]
>>> 2011-06-10 11:09:56.573731 [DEBUG] mod_commands.c:3205
>>> (sofia/external/03996590031055 at 203.208.207.212) State Change
>>> CS_CONSUME_MEDIA -> CS_EXECUTE
>>> 2011-06-10 11:09:56.573731 [DEBUG] switch_core_session.c:1116 Send signal
>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>> 2011-06-10 11:09:56.575262 [DEBUG] switch_core_state_machine.c:320
>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>> CS_EXECUTE
>>> 2011-06-10 11:09:56.575262 [DEBUG] switch_core_state_machine.c:366
>>> (sofia/external/03996590031055 at 203.208.207.212) State EXECUTE
>>> 2011-06-10 11:09:56.575262 [DEBUG] mod_sofia.c:240
>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA EXECUTE
>>> 2011-06-10 11:09:56.575262 [DEBUG] switch_core_state_machine.c:157
>>> sofia/external/03996590031055 at 203.208.207.212 Standard EXECUTE
>>> EXECUTE sofia/external/03996590031055 at 203.208.207.212 park()
>>> 2011-06-10 11:09:56.618770 [DEBUG] switch_rtp.c:2933 Correct ip/port
>>> confirmed.
>>> recv 669 bytes from udp/[203.208.207.212]:5060 at 03:10:01.562021:
>>>
>>> ------------------------------------------------------------------------
>>>   SIP/2.0 183 Session Progress
>>>   Via: SIP/2.0/UDP
>>> 10.1.1.46:5080;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>   To:
>>> &lt;sip:03996590031055 at 203.208.207.212
>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>   CSeq: 13501633 INVITE
>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
>>> SUBSCRIBE, UPDATE
>>>   Content-Type: application/sdp
>>>   Content-Length:   189
>>>
>>>   v=0
>>>   o=- 421265648 2 IN IP4 203.208.207.219
>>>   s=session
>>>   c=IN IP4 203.208.207.196
>>>   t=0 0
>>>   m=audio 30792 RTP/AVP 18 101
>>>   a=rtpmap:18 G729/8000
>>>   a=rtpmap:101 telephone-event/8000
>>>   a=sendrecv
>>>
>>> ------------------------------------------------------------------------
>>> 2011-06-10 11:10:01.455944 [INFO] sofia.c:729
>>> sofia/external/03996590031055 at 203.208.207.212 Update Callee ID to
>>> "03996590031055" <03996590031055>
>>> 2011-06-10 11:10:01.455944 [DEBUG] sofia.c:4641 Channel
>>> sofia/external/03996590031055 at 203.208.207.212 skipping state
>>> [proceeding][183]
>>> recv 669 bytes from udp/[203.208.207.212]:5060 at 03:10:01.563178:
>>>
>>> ------------------------------------------------------------------------
>>>   SIP/2.0 183 Session Progress
>>>   Via: SIP/2.0/UDP
>>> 10.1.1.46:5080;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>   To:
>>> &lt;sip:03996590031055 at 203.208.207.212
>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>   CSeq: 13501633 INVITE
>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
>>> SUBSCRIBE, UPDATE
>>>   Content-Type: application/sdp
>>>   Content-Length:   189
>>>
>>>   v=0
>>>   o=- 421265648 3 IN IP4 203.208.207.219
>>>   s=session
>>>   c=IN IP4 203.208.207.196
>>>   t=0 0
>>>   m=audio 30792 RTP/AVP 18 101
>>>   a=rtpmap:18 G729/8000
>>>   a=rtpmap:101 telephone-event/8000
>>>   a=sendrecv
>>>
>>> ------------------------------------------------------------------------
>>> 2011-06-10 11:10:01.457169 [DEBUG] sofia.c:4641 Channel
>>> sofia/external/03996590031055 at 203.208.207.212 skipping state
>>> [proceeding][183]
>>> recv 749 bytes from udp/[203.208.207.212]:5060 at 03:10:27.365284:
>>>
>>> ------------------------------------------------------------------------
>>>   SIP/2.0 200 Ok
>>>   Via: SIP/2.0/UDP
>>> 10.1.1.46:5080;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>   To:
>>> &lt;sip:03996590031055 at 203.208.207.212
>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>   CSeq: 13501633 INVITE
>>>   Contact: <sip:203.208.207.212:5060>
>>>   Allow-Events: refer
>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
>>> SUBSCRIBE, UPDATE
>>>   Content-Type: application/sdp
>>>   Supported: 100rel, timer, replaces
>>>   Content-Length:   189
>>>
>>>   v=0
>>>   o=- 421265648 4 IN IP4 203.208.207.219
>>>   s=session
>>>   c=IN IP4 203.208.207.196
>>>   t=0 0
>>>   m=audio 30792 RTP/AVP 18 101
>>>   a=rtpmap:18 G729/8000
>>>   a=rtpmap:101 telephone-event/8000
>>>   a=sendrecv
>>>
>>> ------------------------------------------------------------------------
>>> 2011-06-10 11:10:27.258092 [DEBUG] sofia.c:4646 Channel
>>> sofia/external/03996590031055 at 203.208.207.212 entering state
>>> [completing][200]
>>> 2011-06-10 11:10:27.258092 [DEBUG] sofia.c:4657 Remote SDP:
>>> v=0
>>> o=- 421265648 4 IN IP4 203.208.207.219
>>> s=session
>>> c=IN IP4 203.208.207.196
>>> t=0 0
>>> m=audio 30792 RTP/AVP 18 101
>>> a=rtpmap:18 G729/8000
>>> a=rtpmap:101 telephone-event/8000
>>>
>>> send 405 bytes to udp/[203.208.207.212]:5060 at 03:10:27.366562:
>>>
>>> ------------------------------------------------------------------------
>>>   ACK sip:203.208.207.212:5060 SIP/2.0
>>>   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKyUg0ga7pUZmrS
>>>   Max-Forwards: 70
>>>   From: "" &lt;sip:03996563750914 at 202.73.56.46&gt;;tag=H4ZN43UH9S9Qc
>>>   To:
>>> &lt;sip:03996590031055 at 203.208.207.212
>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>   CSeq: 13501633 ACK
>>>   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
>>>   Content-Length: 0
>>>
>>>
>>> ------------------------------------------------------------------------
>>> 2011-06-10 11:10:27.260267 [DEBUG] sofia.c:4646 Channel
>>> sofia/external/03996590031055 at 203.208.207.212 entering state
>>> [ready][200]
>>> 2011-06-10 11:10:27.260267 [DEBUG] switch_channel.c:2782
>>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change EARLY
>>> ->
>>> ACTIVE
>>> 2011-06-10 11:10:27.260267 [NOTICE] sofia.c:5175 Channel
>>> [sofia/external/03996590031055 at 203.208.207.212] has been answered
>>> 2011-06-10 11:10:27.264178 [DEBUG] switch_scheduler.c:214 Added task 23
>>> switch_ivr_schedule_hangup (ea57b74b-a8c2-4fea-9683-98054dc03a79) to run
>>> at
>>> 1307676927
>>> 2011-06-10 11:10:27.265202 [DEBUG] switch_core_session.c:954 Send signal
>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971 variable
>>> string 0 = [origination_caller_id_number=03996590031055]
>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971 variable
>>> string 1 = [ccd_session_id=20110610105829676824]
>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971 variable
>>> string 2 = [sip_cid_type=pid]
>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971 variable
>>> string 3 = [privacy=yes]
>>> 2011-06-10 11:10:27.266218 [NOTICE] switch_channel.c:808 New Channel
>>> sofia/external/03996563750914 at 203.208.207.212
>>> [30228d2b-756a-4a98-871d-db63a2955b52]
>>> 2011-06-10 11:10:27.266218 [DEBUG] mod_sofia.c:4129
>>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_NEW ->
>>> CS_INIT
>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_session.c:1116 Send signal
>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:320
>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>> CS_INIT
>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:356
>>> (sofia/external/03996563750914 at 203.208.207.212) State INIT
>>> 2011-06-10 11:10:27.266218 [DEBUG] mod_sofia.c:84
>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA INIT
>>> 2011-06-10 11:10:27.266218 [DEBUG] mod_sofia.c:124
>>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_INIT ->
>>> CS_ROUTING
>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_session.c:1116 Send signal
>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:356
>>> (sofia/external/03996563750914 at 203.208.207.212) State INIT going to
>>> sleep
>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:320
>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>> CS_ROUTING
>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_channel.c:1657
>>> (sofia/external/03996563750914 at 203.208.207.212) Callstate Change DOWN ->
>>> RINGING
>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:359
>>> (sofia/external/03996563750914 at 203.208.207.212) State ROUTING
>>> send 984 bytes to udp/[203.208.207.212]:5060 at 03:10:27.373220:
>>>
>>> ------------------------------------------------------------------------
>>>   INVITE sip:03996563750914 at 203.208.207.212 SIP/2.0
>>>   Via: SIP/2.0/UDP
>>> 202.73.56.46:5080;rport;branch=z9hG4bKZ49rj5Qtr8aBN2011-06-10
>>> 11:10:27.266218 [DEBUG] mod_sofia.c:147
>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA ROUTING
>>>
>>>   Max-Forwards: 70
>>>   From: "" &lt;sip:03996590031055 at 202.73.56.46&gt;;tag=jDSe6ycN62Zar
>>>   To: &lt;sip:03996563750914 at 203.208.207.212&gt;
>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>   CSeq: 13501649 INVITE
>>>   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
>>>   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>> REGISTER, REFER, NOTIFY
>>>   Supported: timer, precondition, path, replaces
>>>   Allow-Events: talk, hold, refer
>>>   Privacy: none
>>>   Content-Type: application/sdp
>>>   Content-Disposition: session
>>>   Content-Length: 204
>>>   X-FS-Support: update_display
>>>   P-Asserted-Identity: "" &lt;sip:03996590031055 at 202.73.56.46&gt;
>>>
>>>   v=0
>>>   o=FreeSWITCH 1307646863 1307646864 IN IP4 202.73.56.46
>>>   s=FreeSWITCH
>>>   c=IN IP4 202.73.56.46
>>>   t=0 0
>>>   m=audio 28564 RTP/AVP 18 3 101 13
>>>   a=rtpmap:101 telephone-event/8000
>>>   a=fmtp:101 0-16
>>>   a=ptime:20
>>>
>>> ------------------------------------------------------------------------
>>> 2011-06-10 11:10:27.267630 [DEBUG] sofia.c:4646 Channel
>>> sofia/external/03996563750914 at 203.208.207.212 entering state
>>> [calling][0]
>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_ivr_originate.c:66
>>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_ROUTING
>>> ->
>>> CS_CONSUME_MEDIA
>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_session.c:1116 Send signal
>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:359
>>> (sofia/external/03996563750914 at 203.208.207.212) State ROUTING going to
>>> sleep
>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:320
>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>> CS_CONSUME_MEDIA
>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:378
>>> (sofia/external/03996563750914 at 203.208.207.212) State CONSUME_MEDIA
>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:378
>>> (sofia/external/03996563750914 at 203.208.207.212) State CONSUME_MEDIA
>>> going to
>>> sleep
>>> recv 307 bytes from udp/[203.208.207.212]:5060 at 03:10:27.378710:
>>>
>>> ------------------------------------------------------------------------
>>>   SIP/2.0 100 Trying
>>>   Via: SIP/2.0/UDP
>>> 10.1.1.46:5080;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>>   To: &lt;sip:03996563750914 at 203.208.207.212&gt;
>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>   CSeq: 13501649 INVITE
>>>   Content-Length:     0
>>>
>>>
>>> ------------------------------------------------------------------------
>>> 2011-06-10 11:10:27.278484 [DEBUG] switch_ivr.c:563
>>> sofia/external/03996590031055 at 203.208.207.212 Command Execute
>>> playback(/usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav)
>>> EXECUTE sofia/external/03996590031055 at 203.208.207.212
>>> playback(/usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav)
>>> 2011-06-10 11:10:27.278484 [DEBUG] switch_core_file.c:176 File
>>> /usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav sample rate
>>> 11025
>>> doesn't match requested rate 8000
>>> 2011-06-10 11:10:27.278484 [WARNING] switch_core_file.c:189 File has 2
>>> channels, muxing to mono will occur.
>>> 2011-06-10 11:10:27.278484 [DEBUG] switch_ivr_play_say.c:1244 Codec
>>> Activated L16 at 8000hz 2 channels 20ms
>>> 2011-06-10 11:10:27.298764 [INFO] mod_com_g729.c:119 ENCODER CREATE -
>>> 0x2aaab00310c0 0x2aaab00b20c0
>>> recv 666 bytes from udp/[203.208.207.212]:5060 at 03:10:28.618472:
>>>
>>> ------------------------------------------------------------------------
>>>   SIP/2.0 183 Session Progress
>>>   Via: SIP/2.0/UDP
>>> 10.1.1.46:5080;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>>   To:
>>> &lt;sip:03996563750914 at 203.208.207.212
>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>   CSeq: 13501649 INVITE
>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
>>> SUBSCRIBE, UPDATE
>>>   Content-Type: application/sdp
>>>   Content-Length:   186
>>>
>>>   v=0
>>>   o=- 131082 1 IN IP4 203.208.207.218
>>>   s=session
>>>   c=IN IP4 203.208.207.195
>>>   t=0 0
>>>   m=audio 45002 RTP/AVP 18 101
>>>   a=rtpmap:18 G729/8000
>>>   a=rtpmap:101 telephone-event/8000
>>>   a=sendrecv
>>>
>>> ------------------------------------------------------------------------
>>> 2011-06-10 11:10:28.513195 [INFO] sofia.c:729
>>> sofia/external/03996563750914 at 203.208.207.212 Update Callee ID to
>>> "Outbound
>>> Call" <03996563750914>
>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia.c:4646 Channel
>>> sofia/external/03996563750914 at 203.208.207.212 entering state
>>> [proceeding][183]
>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia.c:4657 Remote SDP:
>>> v=0
>>> o=- 131082 1 IN IP4 203.208.207.218
>>> s=session
>>> c=IN IP4 203.208.207.195
>>> t=0 0
>>> m=audio 45002 RTP/AVP 18 101
>>> a=rtpmap:18 G729/8000
>>> a=rtpmap:101 telephone-event/8000
>>>
>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:4467 Audio Codec Compare
>>> [G729:18:8000:20:8000]/[G729:18:8000:20:8000]
>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:2757 Set Codec
>>> sofia/external/03996563750914 at 203.208.207.212 G729/8000 20 ms 160
>>> samples
>>> 8000 bits
>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:4565 Set 2833 dtmf send
>>> payload to 101
>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:2987 AUDIO RTP
>>> [sofia/external/03996563750914 at 203.208.207.212] 10.1.1.46 port 28564 ->
>>> 203.208.207.195 port 45002 codec: 18 ms: 20
>>> 2011-06-10 11:10:28.513195 [DEBUG] switch_rtp.c:1607 Starting timer
>>> [soft]
>>> 160 bytes per 20ms
>>> 2011-06-10 11:10:28.514938 [DEBUG] sofia_glue.c:3228 Set 2833 dtmf send
>>> payload to 101
>>> 2011-06-10 11:10:28.514938 [DEBUG] sofia_glue.c:3233 Set 2833 dtmf
>>> receive
>>> payload to 101
>>> 2011-06-10 11:10:28.514938 [NOTICE] sofia_glue.c:3680 Pre-Answer
>>> sofia/external/03996563750914 at 203.208.207.212!
>>> 2011-06-10 11:10:28.514938 [DEBUG] switch_channel.c:2627
>>> (sofia/external/03996563750914 at 203.208.207.212) Callstate Change RINGING
>>> ->
>>> EARLY
>>> 2011-06-10 11:10:28.514938 [DEBUG] switch_ivr_originate.c:3408 Originate
>>> Resulted in Success: [sofia/external/03996563750914 at 203.208.207.212]
>>> 2011-06-10 11:10:28.514938 [DEBUG] mod_commands.c:3205
>>> (sofia/external/03996563750914 at 203.208.207.212) State Change
>>> CS_CONSUME_MEDIA -> CS_EXECUTE
>>> 2011-06-10 11:10:28.514938 [DEBUG] switch_core_session.c:1116 Send signal
>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>> 2011-06-10 11:10:28.515884 [DEBUG] switch_core_state_machine.c:320
>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>> CS_EXECUTE
>>> 2011-06-10 11:10:28.515884 [DEBUG] switch_core_state_machine.c:366
>>> (sofia/external/03996563750914 at 203.208.207.212) State EXECUTE
>>> 2011-06-10 11:10:28.515884 [DEBUG] mod_sofia.c:240
>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA EXECUTE
>>> 2011-06-10 11:10:28.515884 [DEBUG] switch_core_state_machine.c:157
>>> sofia/external/03996563750914 at 203.208.207.212 Standard EXECUTE
>>> EXECUTE sofia/external/03996563750914 at 203.208.207.212 park()
>>> 2011-06-10 11:10:28.516887 [DEBUG] switch_core_session.c:954 Send signal
>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_ivr_bridge.c:1480
>>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_EXECUTE
>>> ->
>>> CS_HIBERNATE
>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:1116 Send signal
>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_ivr_bridge.c:1482
>>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_EXECUTE
>>> ->
>>> CS_HIBERNATE
>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:1116 Send signal
>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:771 Send signal
>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:771 Send signal
>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_ivr_play_say.c:1581 done
>>> playing
>>> file
>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_ivr.c:563
>>> sofia/external/03996590031055 at 203.208.207.212 Command Execute
>>> playback(tone_stream://%(2000,4000,440,480);loops=10)
>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:366
>>> (sofia/external/03996563750914 at 203.208.207.212) State EXECUTE going to
>>> sleep
>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:320
>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>> CS_HIBERNATE
>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:381
>>> (sofia/external/03996563750914 at 203.208.207.212) State HIBERNATE
>>> 2011-06-10 11:10:28.519223 [DEBUG] mod_sofia.c:221
>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA HIBERNATE
>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_ivr_bridge.c:731
>>> (sofia/external/03996563750914 at 203.208.207.212) State Change
>>> CS_HIBERNATE ->
>>> CS_RESET
>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_session.c:1116 Send signal
>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:381
>>> (sofia/external/03996563750914 at 203.208.207.212) State HIBERNATE going to
>>> sleep
>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:320
>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>> CS_RESET
>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:362
>>> (sofia/external/03996563750914 at 203.208.207.212) State RESET
>>> 2011-06-10 11:10:28.519223 [DEBUG] mod_sofia.c:165
>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA RESET
>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_ivr_bridge.c:716
>>> sofia/external/03996563750914 at 203.208.207.212 CUSTOM RESET
>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:66
>>> sofia/external/03996563750914 at 203.208.207.212 Standard RESET
>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:362
>>> (sofia/external/03996563750914 at 203.208.207.212) State RESET going to
>>> sleep
>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_session.c:709 Send signal
>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>> recv 666 bytes from udp/[203.208.207.212]:5060 at 03:10:30.540814:
>>>
>>> ------------------------------------------------------------------------
>>>   SIP/2.0 183 Session Progress
>>>   Via: SIP/2.0/UDP
>>> 10.1.1.46:5080;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>>   To:
>>> &lt;sip:03996563750914 at 203.208.207.212
>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>   CSeq: 13501649 INVITE
>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
>>> SUBSCRIBE, UPDATE
>>>   Content-Type: application/sdp
>>>   Content-Length:   186
>>>
>>>   v=0
>>>   o=- 131082 2 IN IP4 203.208.207.218
>>>   s=session
>>>   c=IN IP4 203.208.207.195
>>>   t=0 0
>>>   m=audio 45002 RTP/AVP 18 101
>>>   a=rtpmap:18 G729/8000
>>>   a=rtpmap:101 telephone-event/8000
>>>   a=sendrecv
>>>
>>> ------------------------------------------------------------------------
>>> 2011-06-10 11:10:30.435309 [INFO] sofia.c:729
>>> sofia/external/03996563750914 at 203.208.207.212 Update Callee ID to
>>> "03996563750914" <03996563750914>
>>> 2011-06-10 11:10:30.435309 [DEBUG] sofia.c:4641 Channel
>>> sofia/external/03996563750914 at 203.208.207.212 skipping state
>>> [proceeding][183]
>>> 2011-06-10 11:10:30.691404 [WARNING] switch_core_session.c:1940 Cannot
>>> execute app 'playback' media required on an outbound channel that does
>>> not
>>> have media established
>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:366
>>> (sofia/external/03996590031055 at 203.208.207.212) State EXECUTE going to
>>> sleep
>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>> CS_HIBERNATE
>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:381
>>> (sofia/external/03996590031055 at 203.208.207.212) State HIBERNATE
>>> 2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:221
>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA HIBERNATE
>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:731
>>> (sofia/external/03996590031055 at 203.208.207.212) State Change
>>> CS_HIBERNATE ->
>>> CS_RESET
>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_session.c:1116 Send signal
>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:381
>>> (sofia/external/03996590031055 at 203.208.207.212) State HIBERNATE going to
>>> sleep
>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>> CS_RESET
>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:362
>>> (sofia/external/03996590031055 at 203.208.207.212) State RESET
>>> 2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:165
>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA RESET
>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:716
>>> sofia/external/03996590031055 at 203.208.207.212 CUSTOM RESET
>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:723
>>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_RESET ->
>>> CS_SOFT_EXECUTE
>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_session.c:1116 Send signal
>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:362
>>> (sofia/external/03996590031055 at 203.208.207.212) State RESET going to
>>> sleep
>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>> CS_SOFT_EXECUTE
>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:372
>>> (sofia/external/03996590031055 at 203.208.207.212) State SOFT_EXECUTE
>>> 2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:558 SOFIA SOFT_EXECUTE
>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:741
>>> sofia/external/03996590031055 at 203.208.207.212 CUSTOM SOFT_EXECUTE
>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:761
>>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_RESET ->
>>> CS_SOFT_EXECUTE
>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_session.c:1116 Send signal
>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>> CS_SOFT_EXECUTE
>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:372
>>> (sofia/external/03996563750914 at 203.208.207.212) State SOFT_EXECUTE
>>> 2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:558 SOFIA SOFT_EXECUTE
>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:741
>>> sofia/external/03996563750914 at 203.208.207.212 CUSTOM SOFT_EXECUTE
>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:204
>>> sofia/external/03996563750914 at 203.208.207.212 Standard SOFT_EXECUTE
>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:372
>>> (sofia/external/03996563750914 at 203.208.207.212) State SOFT_EXECUTE going
>>> to
>>> sleep
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2535
>>> (sofia/external/03996563750914 at 203.208.207.212) Callstate Change EARLY
>>> ->
>>> HANGUP
>>> 2011-06-10 11:10:30.711798 [NOTICE] switch_ivr_originate.c:1045 Hangup
>>> sofia/external/03996563750914 at 203.208.207.212 [CS_SOFT_EXECUTE]
>>> [DESTINATION_OUT_OF_ORDER]
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2551 Send signal
>>> sofia/external/03996563750914 at 203.208.207.212 [KILL]
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send signal
>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>> CS_HANGUP
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2535
>>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change ACTIVE
>>> ->
>>> HANGUP
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
>>> (sofia/external/03996563750914 at 203.208.207.212) State HANGUP
>>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:457 Channel
>>> sofia/external/03996563750914 at 203.208.207.212 hanging up, cause:
>>> DESTINATION_OUT_OF_ORDER
>>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:510 Sending CANCEL to
>>> sofia/external/03996563750914 at 203.208.207.212
>>> send 390 bytes to udp/[203.208.207.212]:5060 at 03:10:30.818391:
>>>
>>> ------------------------------------------------------------------------
>>>   CANCEL sip:03996563750914 at 203.208.207.212 SIP/2.0
>>>   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKZ49rj5Qtr8aBN
>>>   Max-Forwards: 70
>>>   From: "" &lt;sip:03996590031055 at 202.73.56.46&gt;;tag=jDSe6ycN62Zar
>>>   To: &lt;sip:03996563750914 at 203.208.207.212&gt;
>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>   CSeq: 13501649 CANCEL
>>>   Reason: Q.850;cause=27;text="DESTINATION_OUT_OF_ORDER"
>>>   Content-Length: 0
>>>
>>>
>>> ------------------------------------------------------------------------
>>> 2011-06-10 11:10:30.711798 [NOTICE] switch_ivr_bridge.c:772 Hangup
>>> sofia/external/03996590031055 at 203.208.207.212 [CS_SOFT_EXECUTE]
>>> [ORIGINATOR_CANCEL]
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:46
>>> sofia/external/03996563750914 at 203.208.207.212 Standard HANGUP, cause:
>>> DESTINATION_OUT_OF_ORDER
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
>>> (sofia/external/03996563750914 at 203.208.207.212) State HANGUP going to
>>> sleep
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:351
>>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_HANGUP
>>> ->
>>> CS_REPORTING
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send signal
>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>> CS_REPORTING
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:617
>>> (sofia/external/03996563750914 at 203.208.207.212) State REPORTING
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:53
>>> sofia/external/03996563750914 at 203.208.207.212 Standard REPORTING, cause:
>>> DESTINATION_OUT_OF_ORDER
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:617
>>> (sofia/external/03996563750914 at 203.208.207.212) State REPORTING going to
>>> sleep
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2551 Send signal
>>> sofia/external/03996590031055 at 203.208.207.212 [KILL]
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send signal
>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:372
>>> (sofia/external/03996590031055 at 203.208.207.212) State SOFT_EXECUTE going
>>> to
>>> sleep
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>> CS_HANGUP
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:345
>>> (sofia/external/03996563750914 at 203.208.207.212) State Change
>>> CS_REPORTING ->
>>> CS_DESTROY
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send signal
>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1288 Session 40
>>> (sofia/external/03996563750914 at 203.208.207.212) Locked, Waiting on
>>> external
>>> entities
>>> 2011-06-10 11:10:30.711798 [NOTICE] switch_core_session.c:1306 Session 40
>>> (sofia/external/03996563750914 at 203.208.207.212) Ended
>>> 2011-06-10 11:10:30.711798 [NOTICE] switch_core_session.c:1308 Close
>>> Channel
>>> sofia/external/03996563750914 at 203.208.207.212 [CS_DESTROY]
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:449
>>> (sofia/external/03996563750914 at 203.208.207.212) Callstate Change HANGUP
>>> ->
>>> DOWN
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:452
>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>> CS_DESTROY
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:462
>>> (sofia/external/03996563750914 at 203.208.207.212) State DESTROY
>>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:362
>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA DESTROY
>>> 2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
>>> 0x2aaaac013028 (nil)
>>> 2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
>>> 0x2aaaac013028 (nil)
>>> 2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
>>> 0x2aaaac013088 (nil)
>>> 2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
>>> 0x2aaaac013088 (nil)
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:60
>>> sofia/external/03996563750914 at 203.208.207.212 Standard DESTROY
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:462
>>> (sofia/external/03996563750914 at 203.208.207.212) State DESTROY going to
>>> sleep
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
>>> (sofia/external/03996590031055 at 203.208.207.212) State HANGUP
>>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:457 Channel
>>> sofia/external/03996590031055 at 203.208.207.212 hanging up, cause:
>>> ORIGINATOR_CANCEL
>>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:500 Sending BYE to
>>> sofia/external/03996590031055 at 203.208.207.212
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:46
>>> sofia/external/03996590031055 at 203.208.207.212 Standard HANGUP, cause:
>>> ORIGINATOR_CANCEL
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
>>> (sofia/external/03996590031055 at 203.208.207.212) State HANGUP going to
>>> sleep
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:351
>>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_HANGUP
>>> ->
>>> CS_REPORTING
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send signal
>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>> CS_REPORTING
>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:617
>>> (sofia/external/03996590031055 at 203.208.207.212) State REPORTING
>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:53
>>> sofia/external/03996590031055 at 203.208.207.212 Standard REPORTING, cause:
>>> ORIGINATOR_CANCEL
>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:617
>>> (sofia/external/03996590031055 at 203.208.207.212) State REPORTING going to
>>> sleep
>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:345
>>> (sofia/external/03996590031055 at 203.208.207.212) State Change
>>> CS_REPORTING ->
>>> CS_DESTROY
>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_session.c:1116 Send signal
>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_session.c:1288 Session 39
>>> (sofia/external/03996590031055 at 203.208.207.212) Locked, Waiting on
>>> external
>>> entities
>>> 2011-06-10 11:10:30.714001 [NOTICE] switch_core_session.c:1306 Session 39
>>> (sofia/external/03996590031055 at 203.208.207.212) Ended
>>> 2011-06-10 11:10:30.714001 [NOTICE] switch_core_session.c:1308 Close
>>> Channel
>>> sofia/external/03996590031055 at 203.208.207.212 [CS_DESTROY]
>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:449
>>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change HANGUP
>>> ->
>>> DOWN
>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:452
>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>> CS_DESTROY
>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:462
>>> (sofia/external/03996590031055 at 203.208.207.212) State DESTROY
>>> 2011-06-10 11:10:30.714001 [DEBUG] mod_sofia.c:362
>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA DESTROY
>>> 2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
>>> 0x2aaab0031060 (nil)
>>> 2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
>>> 0x2aaab0031060 (nil)
>>> 2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
>>> 0x2aaab00310c0 0x2aaab00b20c0
>>> 2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
>>> 0x2aaab00310c0 (nil)
>>> send 662 bytes to udp/[203.208.207.212]:5060 at 03:10:30.820530:
>>>
>>> ------------------------------------------------------------------------
>>>   BYE sip:203.208.207.212:5060 SIP/2.0
>>>   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bK0D3Hm08XNH1Xg
>>>   Max-Forwards: 70
>>>   From: "" &lt;sip:03996563750914 at 202.73.56.46&gt;;tag=H4ZN43UH9S9Qc
>>>   To:
>>> &lt;sip:03996590031055 at 203.208.207.212
>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>   CSeq: 13501634 BYE
>>>   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
>>>   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>> REGISTER, REFER, NOTIFY
>>>   Supported: timer, precondition, path, replaces
>>>   Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL"
>>>   Content-Length: 0
>>>
>>>
>>> ------------------------------------------------------------------------
>>> 2011-06-10 11:10:30.715878 [INFO] mod_com_g729.c:83 ENCODER DESTROY -
>>> 0x2aaab00310c0 0x2aaab00b20c0
>>> 2011-06-10 11:10:30.715878 [DEBUG] switch_core_state_machine.c:60
>>> sofia/external/03996590031055 at 203.208.207.212 Standard DESTROY
>>> 2011-06-10 11:10:30.715878 [DEBUG] switch_core_state_machine.c:462
>>> (sofia/external/03996590031055 at 203.208.207.212) State DESTROY going to
>>> sleep
>>> recv 383 bytes from udp/[203.208.207.212]:5060 at 03:10:30.823302:
>>>
>>> ------------------------------------------------------------------------
>>>   SIP/2.0 200 Ok
>>>   Via: SIP/2.0/UDP
>>> 10.1.1.46:5080;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>>   To:
>>> &lt;sip:03996563750914 at 203.208.207.212
>>> &gt;;tag=2QGB951HCR30000E1D00000u00000001QXU3LU
>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>   CSeq: 13501649 CANCEL
>>>   Contact: <sip:203.208.207.212:5060>
>>>   Content-Length:     0
>>>
>>>
>>> ------------------------------------------------------------------------
>>> recv 411 bytes from udp/[203.208.207.212]:5060 at 03:10:30.824765:
>>>
>>> ------------------------------------------------------------------------
>>>   SIP/2.0 487 Request Terminated
>>>   Via: SIP/2.0/UDP
>>> 10.1.1.46:5080;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>>   To:
>>> &lt;sip:03996563750914 at 203.208.207.212
>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>   CSeq: 13501649 INVITE
>>>   Reason: SIP;cause=487;text="Request Terminated"
>>>   Content-Length:     0
>>>
>>>
>>> ------------------------------------------------------------------------
>>> send 371 bytes to udp/[203.208.207.212]:5060 at 03:10:30.824891:
>>>
>>> ------------------------------------------------------------------------
>>>   ACK sip:03996563750914 at 203.208.207.212 SIP/2.0
>>>   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKZ49rj5Qtr8aBN
>>>   Max-Forwards: 70
>>>   From: "" &lt;sip:03996590031055 at 202.73.56.46&gt;;tag=jDSe6ycN62Zar
>>>   To:
>>> &lt;sip:03996563750914 at 203.208.207.212
>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>   CSeq: 13501649 ACK
>>>   Content-Length: 0
>>>
>>>
>>> ------------------------------------------------------------------------
>>> recv 380 bytes from udp/[203.208.207.212]:5060 at 03:10:30.826375:
>>>
>>> ------------------------------------------------------------------------
>>>   SIP/2.0 200 Ok
>>>   Via: SIP/2.0/UDP
>>> 10.1.1.46:5080;rport=5080;branch=z9hG4bK0D3Hm08XNH1Xg;received=10.1.1.46
>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>   To:
>>> &lt;sip:03996590031055 at 203.208.207.212
>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>   CSeq: 13501634 BYE
>>>
>>>
>>>
>>> --
>>> View this message in context:
>>> http://freeswitch-users.2379917.n2.nabble.com/Leg-A-is-automatically-disconnected-on-Leg-B-orginate-failure-tp6482192p6482192.html
>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From msc at freeswitch.org  Fri Jun 24 10:39:00 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 23 Jun 2011 23:39:00 -0700
Subject: [Freeswitch-users] Any RECORD_STOP event sample available?
In-Reply-To: <BANLkTimkbxVe3GxFfaHY-+LxBs5_umpGSA@mail.gmail.com>
References: <BANLkTimkbxVe3GxFfaHY-+LxBs5_umpGSA@mail.gmail.com>
Message-ID: <BANLkTikeaZMrJkdYOWjROzNX=cvFv9-1qA@mail.gmail.com>

It would be better for you to experience an actual record stop event:

open fs_cli and enter:
/log 0
/event plain event_stop

Then make a call to 9664 (or some other valid extension on your system)
Back at fs_cli:

uuid_record xxx start /tmp/rec.wav
uuid_record xxx stop /tmp/wec.wav

As soon as you issue the stop command your screen will be filled with all
the wonderful stuff in the RECORD_STOP event. I think you'll find the event
headers to be pretty self-explanatory, but if you have any questions just
ask here if you don't see anything on the wiki.

Thanks,
MC

On Thu, Jun 23, 2011 at 7:35 PM, Frankie Yiu <frankie.k.yiu at gmail.com>wrote:

> Hi there,
>
> Could someone please point me where I can find a sample of the RECORD_STOP
> event look like?
> I look at this page but it does not have any information about it. (
> http://wiki.freeswitch.org/wiki/Event_list#RECORD_STOP)
>
> Thanks,
> Frankie
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri Jun 24 10:41:26 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 23 Jun 2011 23:41:26 -0700
Subject: [Freeswitch-users] Leg-A is automatically disconnected on Leg-B
 orginate failure
In-Reply-To: <BANLkTi=m7oMix1ZRC9URtRuAbsG2fWyhMQ@mail.gmail.com>
References: <1308213151543-6482192.post@n2.nabble.com>
	<BANLkTi=xYgBTPVGxVGmk9f2=bmX5X2vF=Q@mail.gmail.com>
	<BANLkTikhPmjN5HeeJHBnQt1ivA8zbtCnYA@mail.gmail.com>
	<BANLkTi=m7oMix1ZRC9URtRuAbsG2fWyhMQ@mail.gmail.com>
Message-ID: <BANLkTimxBU8m311LVKMWcFOjukwLg+ZUGQ@mail.gmail.com>

Pastebin the entire debug log, including the siptrace. Also include the
originate line and any other dialplan config that might be used.
-MC

On Thu, Jun 23, 2011 at 11:26 PM, David Ma <mays.david at gmail.com> wrote:

> Hi Michael,
>
> Unfortunately this problem still happens.
>
> I enabled "continue_on_fail" for leg-A when I originated the call. Leg-A
> call went well. Then I originated leg-B call ("continue_on_fail" is NOT set
> for leg-B), which failed for [DESTINATION_OUT_OF_ORDER]. As the consequence,
> leg-A was hung up by FS automatically for [ORIGINATOR_CANCEL].
>
> The log excerpt follows.
>
> Do you think "continue_on_fail" should be also enabled for leg-B call?
>
> Thanks,
> D.Ma
>
> 2011-06-24 13:29:35.887830 [DEBUG] switch_ivr_originate.c:1971 variable
> string 5 = [continue_on_fail=true]
>
> 2011-06-24 13:29:35.887830 [NOTICE] switch_channel.c:808 New Channel
> sofia/external/03996597632298 at 203.208.207.212[c0bd700d-913c-42ad-b68f-81001bf658b8]
> [...]
> 2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:372
> (sofia/external/03996563750911 at 203.208.207.212) State SOFT_EXECUTE going
> to sleep
> 2011-06-24 13:29:48.161779 [DEBUG] switch_channel.c:2535 (sofia/external/
> 03996563750911 at 203.208.207.212) Callstate Change EARLY -> HANGUP
> 2011-06-24 13:29:48.161779 [NOTICE] switch_ivr_originate.c:1045 Hangup
> sofia/external/03996563750911 at 203.208.207.212 [CS_SOFT_EXECUTE]
> [DESTINATION_OUT_OF_ORDER]
> 2011-06-24 13:29:48.161779 [DEBUG] switch_channel.c:2551 Send signal
> sofia/external/03996563750911 at 203.208.207.212 [KILL]
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996563750911 at 203.208.207.212 [BREAK]
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996563750911 at 203.208.207.212) Running State Change
> CS_HANGUP
> 2011-06-24 13:29:48.161779 [DEBUG] switch_channel.c:2535 (sofia/external/
> 03996597632298 at 203.208.207.212) Callstate Change ACTIVE -> HANGUP
> 2011-06-24 13:29:48.161779 [NOTICE] switch_ivr_bridge.c:772 Hangup
> sofia/external/03996597632298 at 203.208.207.212 [CS_SOFT_EXECUTE]
> [ORIGINATOR_CANCEL]
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:557
> (sofia/external/03996563750911 at 203.208.207.212) State HANGUP
> 2011-06-24 13:29:48.161779 [DEBUG] mod_sofia.c:457 Channel sofia/external/
> 03996563750911 at 203.208.207.212 hanging up, cause: DESTINATION_OUT_OF_ORDER
> 2011-06-24 13:29:48.161779 [DEBUG] mod_sofia.c:510 Sending CANCEL to
> sofia/external/03996563750911 at 203.208.207.212
> 2011-06-24 13:29:48.161779 [DEBUG] switch_channel.c:2551 Send signal
> sofia/external/03996597632298 at 203.208.207.212 [KILL]
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996597632298 at 203.208.207.212 [BREAK]
>
> On Fri, Jun 17, 2011 at 10:52 AM, David Ma <mays.david at gmail.com> wrote:
>
>> Hi Michael,
>>
>> Thanks very much for your prompt response! I appreciate the information
>> provided.
>>
>> I was actually searching the the existence of such a variable. I was not
>> so luck to find it out and thereby resort to the support forum.
>>
>> I've modified my code to build this parameter into my application. Will
>> feedback to you after verification.
>>
>> Thanks again,
>> D.Ma
>>
>> On Fri, Jun 17, 2011 at 4:51 AM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> How about setting this?
>>> http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
>>>
>>> -MC
>>>
>>>
>>> On Thu, Jun 16, 2011 at 1:32 AM, dma <mays.david at gmail.com> wrote:
>>>
>>>> I am creating a call-back solution. After leg-A answers, I originate
>>>> leg-B
>>>> call. After receiving SIP 183 from Leg-B, I bridge the 2 legs. However,
>>>> in
>>>> some cases, leg-A is automatically disconnected by FreeSwitch on leg-B
>>>> failure, for example, DESTINATION_OUT_OF_ORDER. The application is not
>>>> given
>>>> a chance to handle leg-B failure event. This should not be a correct
>>>> scenario because I never set "hangup-after-bridge", which is false by
>>>> default.
>>>>
>>>> The right way should be, FreeSwitch doesn't hang up leg-A automatically,
>>>> but
>>>> give a chance for the application to decide what to do.
>>>>
>>>> Please see the logs below:
>>>>
>>>> =================================================
>>>>
>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
>>>> string 0 = [origination_caller_id_number=03996563750914]
>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
>>>> string 1 = [originate_timeout=30]
>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
>>>> string 2 = [ccd_session_id=20110610105829676824]
>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
>>>> string 3 = [sip_cid_type=pid]
>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
>>>> string 4 = [privacy=yes]
>>>> 2011-06-10 11:09:55.370506 [NOTICE] switch_channel.c:808 New Channel
>>>> sofia/external/03996590031055 at 203.208.207.212
>>>> [ea57b74b-a8c2-4fea-9683-98054dc03a79]
>>>> 2011-06-10 11:09:55.370506 [DEBUG] mod_sofia.c:4129
>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_NEW ->
>>>> CS_INIT
>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_core_session.c:1116 Send
>>>> signal
>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_core_state_machine.c:320
>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>> CS_INIT
>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_core_state_machine.c:356
>>>> (sofia/external/03996590031055 at 203.208.207.212) State INIT
>>>> 2011-06-10 11:09:55.370506 [DEBUG] mod_sofia.c:84
>>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA INIT
>>>> send 984 bytes to udp/[203.208.207.212]:5060 at 03:09:55.477997:
>>>>
>>>> ------------------------------------------------------------------------
>>>>   INVITE sip:03996590031055 at 203.208.207.212 SIP/2.0
>>>>   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKXjQ7eFpKypy5D
>>>>   Max-Forwards: 70
>>>>   From: "" &lt;sip:03996563750914 at 202.73.56.46&gt;;tag=H4ZN43UH9S9Qc
>>>>   To: &lt;sip:03996590031055 at 203.208.207.212&gt;
>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>   CSeq: 13501633 INVITE
>>>>   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
>>>>   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>> REGISTER, REFER, NOTIFY
>>>>   Supported: timer, precondition, path, replaces
>>>>   Allow-Events: talk, hold, refer
>>>>   Privacy: none
>>>>   Content-Type: application/sdp
>>>>   Content-Disposition: session
>>>>   Content-Length: 204
>>>>   X-FS-Support: update_display
>>>>   P-Asserted-Identity: "" &lt;sip:03996563750914 at 202.73.56.46&gt;
>>>>
>>>>   v=0
>>>>   o=FreeSWITCH 1307645395 1307645396 IN IP4 202.73.56.46
>>>>   s=FreeSWITCH
>>>>   c=IN IP4 202.73.56.46
>>>>   t=0 0
>>>>   m=audio 30000 RTP/AVP 18 3 101 13
>>>>   a=rtpmap:101 telephone-event/8000
>>>>   a=fmtp:101 0-16
>>>>   a=ptime:20
>>>>
>>>> ------------------------------------------------------------------------
>>>> 2011-06-10 11:09:55.371674 [DEBUG] mod_sofia.c:124
>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_INIT ->
>>>> CS_ROUTING
>>>> 2011-06-10 11:09:55.371674 [DEBUG] switch_core_session.c:1116 Send
>>>> signal
>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>> 2011-06-10 11:09:55.371674 [DEBUG] sofia.c:4646 Channel
>>>> sofia/external/03996590031055 at 203.208.207.212 entering state
>>>> [calling][0]
>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:356
>>>> (sofia/external/03996590031055 at 203.208.207.212) State INIT going to
>>>> sleep
>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:320
>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>> CS_ROUTING
>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_channel.c:1657
>>>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change DOWN
>>>> ->
>>>> RINGING
>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:359
>>>> (sofia/external/03996590031055 at 203.208.207.212) State ROUTING
>>>> 2011-06-10 11:09:55.373478 [DEBUG] mod_sofia.c:147
>>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA ROUTING
>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_ivr_originate.c:66
>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_ROUTING
>>>> ->
>>>> CS_CONSUME_MEDIA
>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_session.c:1116 Send
>>>> signal
>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:359
>>>> (sofia/external/03996590031055 at 203.208.207.212) State ROUTING going to
>>>> sleep
>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:320
>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>> CS_CONSUME_MEDIA
>>>> 2011-06-10 11:09:55.374485 [DEBUG] switch_core_state_machine.c:378
>>>> (sofia/external/03996590031055 at 203.208.207.212) State CONSUME_MEDIA
>>>> 2011-06-10 11:09:55.374485 [DEBUG] switch_core_state_machine.c:378
>>>> (sofia/external/03996590031055 at 203.208.207.212) State CONSUME_MEDIA
>>>> going to
>>>> sleep
>>>> recv 307 bytes from udp/[203.208.207.212]:5060 at 03:09:55.485130:
>>>>
>>>> ------------------------------------------------------------------------
>>>>   SIP/2.0 100 Trying
>>>>   Via: SIP/2.0/UDP
>>>> 10.1.1.46:5080
>>>> ;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>>   To: &lt;sip:03996590031055 at 203.208.207.212&gt;
>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>   CSeq: 13501633 INVITE
>>>>   Content-Length:     0
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>> recv 669 bytes from udp/[203.208.207.212]:5060 at 03:09:56.677788:
>>>>
>>>> ------------------------------------------------------------------------
>>>>   SIP/2.0 183 Session Progress
>>>>   Via: SIP/2.0/UDP
>>>> 10.1.1.46:5080
>>>> ;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>>   To:
>>>> &lt;sip:03996590031055 at 203.208.207.212
>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>   CSeq: 13501633 INVITE
>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
>>>> SUBSCRIBE, UPDATE
>>>>   Content-Type: application/sdp
>>>>   Content-Length:   189
>>>>
>>>>   v=0
>>>>   o=- 421265648 1 IN IP4 203.208.207.219
>>>>   s=session
>>>>   c=IN IP4 203.208.207.196
>>>>   t=0 0
>>>>   m=audio 30792 RTP/AVP 18 101
>>>>   a=rtpmap:18 G729/8000
>>>>   a=rtpmap:101 telephone-event/8000
>>>>   a=sendrecv
>>>>
>>>> ------------------------------------------------------------------------
>>>> 2011-06-10 11:09:56.571839 [INFO] sofia.c:729
>>>> sofia/external/03996590031055 at 203.208.207.212 Update Callee ID to
>>>> "Outbound
>>>> Call" <03996590031055>
>>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia.c:4646 Channel
>>>> sofia/external/03996590031055 at 203.208.207.212 entering state
>>>> [proceeding][183]
>>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia.c:4657 Remote SDP:
>>>> v=0
>>>> o=- 421265648 1 IN IP4 203.208.207.219
>>>> s=session
>>>> c=IN IP4 203.208.207.196
>>>> t=0 0
>>>> m=audio 30792 RTP/AVP 18 101
>>>> a=rtpmap:18 G729/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>>
>>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:4467 Audio Codec Compare
>>>> [G729:18:8000:20:8000]/[G729:18:8000:20:8000]
>>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:2757 Set Codec
>>>> sofia/external/03996590031055 at 203.208.207.212 G729/8000 20 ms 160
>>>> samples
>>>> 8000 bits
>>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:4565 Set 2833 dtmf send
>>>> payload to 101
>>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:2987 AUDIO RTP
>>>> [sofia/external/03996590031055 at 203.208.207.212] 10.1.1.46 port 30000 ->
>>>> 203.208.207.196 port 30792 codec: 18 ms: 20
>>>> 2011-06-10 11:09:56.571839 [DEBUG] switch_rtp.c:1607 Starting timer
>>>> [soft]
>>>> 160 bytes per 20ms
>>>> 2011-06-10 11:09:56.573731 [DEBUG] sofia_glue.c:3228 Set 2833 dtmf send
>>>> payload to 101
>>>> 2011-06-10 11:09:56.573731 [DEBUG] sofia_glue.c:3233 Set 2833 dtmf
>>>> receive
>>>> payload to 101
>>>> 2011-06-10 11:09:56.573731 [NOTICE] sofia_glue.c:3680 Pre-Answer
>>>> sofia/external/03996590031055 at 203.208.207.212!
>>>> 2011-06-10 11:09:56.573731 [DEBUG] switch_channel.c:2627
>>>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change
>>>> RINGING ->
>>>> EARLY
>>>> 2011-06-10 11:09:56.573731 [DEBUG] switch_ivr_originate.c:3408 Originate
>>>> Resulted in Success: [sofia/external/03996590031055 at 203.208.207.212]
>>>> 2011-06-10 11:09:56.573731 [DEBUG] mod_commands.c:3205
>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change
>>>> CS_CONSUME_MEDIA -> CS_EXECUTE
>>>> 2011-06-10 11:09:56.573731 [DEBUG] switch_core_session.c:1116 Send
>>>> signal
>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>> 2011-06-10 11:09:56.575262 [DEBUG] switch_core_state_machine.c:320
>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>> CS_EXECUTE
>>>> 2011-06-10 11:09:56.575262 [DEBUG] switch_core_state_machine.c:366
>>>> (sofia/external/03996590031055 at 203.208.207.212) State EXECUTE
>>>> 2011-06-10 11:09:56.575262 [DEBUG] mod_sofia.c:240
>>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA EXECUTE
>>>> 2011-06-10 11:09:56.575262 [DEBUG] switch_core_state_machine.c:157
>>>> sofia/external/03996590031055 at 203.208.207.212 Standard EXECUTE
>>>> EXECUTE sofia/external/03996590031055 at 203.208.207.212 park()
>>>> 2011-06-10 11:09:56.618770 [DEBUG] switch_rtp.c:2933 Correct ip/port
>>>> confirmed.
>>>> recv 669 bytes from udp/[203.208.207.212]:5060 at 03:10:01.562021:
>>>>
>>>> ------------------------------------------------------------------------
>>>>   SIP/2.0 183 Session Progress
>>>>   Via: SIP/2.0/UDP
>>>> 10.1.1.46:5080
>>>> ;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>>   To:
>>>> &lt;sip:03996590031055 at 203.208.207.212
>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>   CSeq: 13501633 INVITE
>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
>>>> SUBSCRIBE, UPDATE
>>>>   Content-Type: application/sdp
>>>>   Content-Length:   189
>>>>
>>>>   v=0
>>>>   o=- 421265648 2 IN IP4 203.208.207.219
>>>>   s=session
>>>>   c=IN IP4 203.208.207.196
>>>>   t=0 0
>>>>   m=audio 30792 RTP/AVP 18 101
>>>>   a=rtpmap:18 G729/8000
>>>>   a=rtpmap:101 telephone-event/8000
>>>>   a=sendrecv
>>>>
>>>> ------------------------------------------------------------------------
>>>> 2011-06-10 11:10:01.455944 [INFO] sofia.c:729
>>>> sofia/external/03996590031055 at 203.208.207.212 Update Callee ID to
>>>> "03996590031055" <03996590031055>
>>>> 2011-06-10 11:10:01.455944 [DEBUG] sofia.c:4641 Channel
>>>> sofia/external/03996590031055 at 203.208.207.212 skipping state
>>>> [proceeding][183]
>>>> recv 669 bytes from udp/[203.208.207.212]:5060 at 03:10:01.563178:
>>>>
>>>> ------------------------------------------------------------------------
>>>>   SIP/2.0 183 Session Progress
>>>>   Via: SIP/2.0/UDP
>>>> 10.1.1.46:5080
>>>> ;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>>   To:
>>>> &lt;sip:03996590031055 at 203.208.207.212
>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>   CSeq: 13501633 INVITE
>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
>>>> SUBSCRIBE, UPDATE
>>>>   Content-Type: application/sdp
>>>>   Content-Length:   189
>>>>
>>>>   v=0
>>>>   o=- 421265648 3 IN IP4 203.208.207.219
>>>>   s=session
>>>>   c=IN IP4 203.208.207.196
>>>>   t=0 0
>>>>   m=audio 30792 RTP/AVP 18 101
>>>>   a=rtpmap:18 G729/8000
>>>>   a=rtpmap:101 telephone-event/8000
>>>>   a=sendrecv
>>>>
>>>> ------------------------------------------------------------------------
>>>> 2011-06-10 11:10:01.457169 [DEBUG] sofia.c:4641 Channel
>>>> sofia/external/03996590031055 at 203.208.207.212 skipping state
>>>> [proceeding][183]
>>>> recv 749 bytes from udp/[203.208.207.212]:5060 at 03:10:27.365284:
>>>>
>>>> ------------------------------------------------------------------------
>>>>   SIP/2.0 200 Ok
>>>>   Via: SIP/2.0/UDP
>>>> 10.1.1.46:5080
>>>> ;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>>   To:
>>>> &lt;sip:03996590031055 at 203.208.207.212
>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>   CSeq: 13501633 INVITE
>>>>   Contact: <sip:203.208.207.212:5060>
>>>>   Allow-Events: refer
>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
>>>> SUBSCRIBE, UPDATE
>>>>   Content-Type: application/sdp
>>>>   Supported: 100rel, timer, replaces
>>>>   Content-Length:   189
>>>>
>>>>   v=0
>>>>   o=- 421265648 4 IN IP4 203.208.207.219
>>>>   s=session
>>>>   c=IN IP4 203.208.207.196
>>>>   t=0 0
>>>>   m=audio 30792 RTP/AVP 18 101
>>>>   a=rtpmap:18 G729/8000
>>>>   a=rtpmap:101 telephone-event/8000
>>>>   a=sendrecv
>>>>
>>>> ------------------------------------------------------------------------
>>>> 2011-06-10 11:10:27.258092 [DEBUG] sofia.c:4646 Channel
>>>> sofia/external/03996590031055 at 203.208.207.212 entering state
>>>> [completing][200]
>>>> 2011-06-10 11:10:27.258092 [DEBUG] sofia.c:4657 Remote SDP:
>>>> v=0
>>>> o=- 421265648 4 IN IP4 203.208.207.219
>>>> s=session
>>>> c=IN IP4 203.208.207.196
>>>> t=0 0
>>>> m=audio 30792 RTP/AVP 18 101
>>>> a=rtpmap:18 G729/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>>
>>>> send 405 bytes to udp/[203.208.207.212]:5060 at 03:10:27.366562:
>>>>
>>>> ------------------------------------------------------------------------
>>>>   ACK sip:203.208.207.212:5060 SIP/2.0
>>>>   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKyUg0ga7pUZmrS
>>>>   Max-Forwards: 70
>>>>   From: "" &lt;sip:03996563750914 at 202.73.56.46&gt;;tag=H4ZN43UH9S9Qc
>>>>   To:
>>>> &lt;sip:03996590031055 at 203.208.207.212
>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>   CSeq: 13501633 ACK
>>>>   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
>>>>   Content-Length: 0
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>> 2011-06-10 11:10:27.260267 [DEBUG] sofia.c:4646 Channel
>>>> sofia/external/03996590031055 at 203.208.207.212 entering state
>>>> [ready][200]
>>>> 2011-06-10 11:10:27.260267 [DEBUG] switch_channel.c:2782
>>>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change EARLY
>>>> ->
>>>> ACTIVE
>>>> 2011-06-10 11:10:27.260267 [NOTICE] sofia.c:5175 Channel
>>>> [sofia/external/03996590031055 at 203.208.207.212] has been answered
>>>> 2011-06-10 11:10:27.264178 [DEBUG] switch_scheduler.c:214 Added task 23
>>>> switch_ivr_schedule_hangup (ea57b74b-a8c2-4fea-9683-98054dc03a79) to run
>>>> at
>>>> 1307676927
>>>> 2011-06-10 11:10:27.265202 [DEBUG] switch_core_session.c:954 Send signal
>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971 variable
>>>> string 0 = [origination_caller_id_number=03996590031055]
>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971 variable
>>>> string 1 = [ccd_session_id=20110610105829676824]
>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971 variable
>>>> string 2 = [sip_cid_type=pid]
>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971 variable
>>>> string 3 = [privacy=yes]
>>>> 2011-06-10 11:10:27.266218 [NOTICE] switch_channel.c:808 New Channel
>>>> sofia/external/03996563750914 at 203.208.207.212
>>>> [30228d2b-756a-4a98-871d-db63a2955b52]
>>>> 2011-06-10 11:10:27.266218 [DEBUG] mod_sofia.c:4129
>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_NEW ->
>>>> CS_INIT
>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_session.c:1116 Send
>>>> signal
>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:320
>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>> CS_INIT
>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:356
>>>> (sofia/external/03996563750914 at 203.208.207.212) State INIT
>>>> 2011-06-10 11:10:27.266218 [DEBUG] mod_sofia.c:84
>>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA INIT
>>>> 2011-06-10 11:10:27.266218 [DEBUG] mod_sofia.c:124
>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_INIT ->
>>>> CS_ROUTING
>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_session.c:1116 Send
>>>> signal
>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:356
>>>> (sofia/external/03996563750914 at 203.208.207.212) State INIT going to
>>>> sleep
>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:320
>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>> CS_ROUTING
>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_channel.c:1657
>>>> (sofia/external/03996563750914 at 203.208.207.212) Callstate Change DOWN
>>>> ->
>>>> RINGING
>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:359
>>>> (sofia/external/03996563750914 at 203.208.207.212) State ROUTING
>>>> send 984 bytes to udp/[203.208.207.212]:5060 at 03:10:27.373220:
>>>>
>>>> ------------------------------------------------------------------------
>>>>   INVITE sip:03996563750914 at 203.208.207.212 SIP/2.0
>>>>   Via: SIP/2.0/UDP
>>>> 202.73.56.46:5080;rport;branch=z9hG4bKZ49rj5Qtr8aBN2011-06-10
>>>> 11:10:27.266218 [DEBUG] mod_sofia.c:147
>>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA ROUTING
>>>>
>>>>   Max-Forwards: 70
>>>>   From: "" &lt;sip:03996590031055 at 202.73.56.46&gt;;tag=jDSe6ycN62Zar
>>>>   To: &lt;sip:03996563750914 at 203.208.207.212&gt;
>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>   CSeq: 13501649 INVITE
>>>>   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
>>>>   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>> REGISTER, REFER, NOTIFY
>>>>   Supported: timer, precondition, path, replaces
>>>>   Allow-Events: talk, hold, refer
>>>>   Privacy: none
>>>>   Content-Type: application/sdp
>>>>   Content-Disposition: session
>>>>   Content-Length: 204
>>>>   X-FS-Support: update_display
>>>>   P-Asserted-Identity: "" &lt;sip:03996590031055 at 202.73.56.46&gt;
>>>>
>>>>   v=0
>>>>   o=FreeSWITCH 1307646863 1307646864 IN IP4 202.73.56.46
>>>>   s=FreeSWITCH
>>>>   c=IN IP4 202.73.56.46
>>>>   t=0 0
>>>>   m=audio 28564 RTP/AVP 18 3 101 13
>>>>   a=rtpmap:101 telephone-event/8000
>>>>   a=fmtp:101 0-16
>>>>   a=ptime:20
>>>>
>>>> ------------------------------------------------------------------------
>>>> 2011-06-10 11:10:27.267630 [DEBUG] sofia.c:4646 Channel
>>>> sofia/external/03996563750914 at 203.208.207.212 entering state
>>>> [calling][0]
>>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_ivr_originate.c:66
>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_ROUTING
>>>> ->
>>>> CS_CONSUME_MEDIA
>>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_session.c:1116 Send
>>>> signal
>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:359
>>>> (sofia/external/03996563750914 at 203.208.207.212) State ROUTING going to
>>>> sleep
>>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:320
>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>> CS_CONSUME_MEDIA
>>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:378
>>>> (sofia/external/03996563750914 at 203.208.207.212) State CONSUME_MEDIA
>>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:378
>>>> (sofia/external/03996563750914 at 203.208.207.212) State CONSUME_MEDIA
>>>> going to
>>>> sleep
>>>> recv 307 bytes from udp/[203.208.207.212]:5060 at 03:10:27.378710:
>>>>
>>>> ------------------------------------------------------------------------
>>>>   SIP/2.0 100 Trying
>>>>   Via: SIP/2.0/UDP
>>>> 10.1.1.46:5080
>>>> ;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>>>   To: &lt;sip:03996563750914 at 203.208.207.212&gt;
>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>   CSeq: 13501649 INVITE
>>>>   Content-Length:     0
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>> 2011-06-10 11:10:27.278484 [DEBUG] switch_ivr.c:563
>>>> sofia/external/03996590031055 at 203.208.207.212 Command Execute
>>>> playback(/usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav)
>>>> EXECUTE sofia/external/03996590031055 at 203.208.207.212
>>>> playback(/usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav)
>>>> 2011-06-10 11:10:27.278484 [DEBUG] switch_core_file.c:176 File
>>>> /usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav sample rate
>>>> 11025
>>>> doesn't match requested rate 8000
>>>> 2011-06-10 11:10:27.278484 [WARNING] switch_core_file.c:189 File has 2
>>>> channels, muxing to mono will occur.
>>>> 2011-06-10 11:10:27.278484 [DEBUG] switch_ivr_play_say.c:1244 Codec
>>>> Activated L16 at 8000hz 2 channels 20ms
>>>> 2011-06-10 11:10:27.298764 [INFO] mod_com_g729.c:119 ENCODER CREATE -
>>>> 0x2aaab00310c0 0x2aaab00b20c0
>>>> recv 666 bytes from udp/[203.208.207.212]:5060 at 03:10:28.618472:
>>>>
>>>> ------------------------------------------------------------------------
>>>>   SIP/2.0 183 Session Progress
>>>>   Via: SIP/2.0/UDP
>>>> 10.1.1.46:5080
>>>> ;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>>>   To:
>>>> &lt;sip:03996563750914 at 203.208.207.212
>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>   CSeq: 13501649 INVITE
>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
>>>> SUBSCRIBE, UPDATE
>>>>   Content-Type: application/sdp
>>>>   Content-Length:   186
>>>>
>>>>   v=0
>>>>   o=- 131082 1 IN IP4 203.208.207.218
>>>>   s=session
>>>>   c=IN IP4 203.208.207.195
>>>>   t=0 0
>>>>   m=audio 45002 RTP/AVP 18 101
>>>>   a=rtpmap:18 G729/8000
>>>>   a=rtpmap:101 telephone-event/8000
>>>>   a=sendrecv
>>>>
>>>> ------------------------------------------------------------------------
>>>> 2011-06-10 11:10:28.513195 [INFO] sofia.c:729
>>>> sofia/external/03996563750914 at 203.208.207.212 Update Callee ID to
>>>> "Outbound
>>>> Call" <03996563750914>
>>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia.c:4646 Channel
>>>> sofia/external/03996563750914 at 203.208.207.212 entering state
>>>> [proceeding][183]
>>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia.c:4657 Remote SDP:
>>>> v=0
>>>> o=- 131082 1 IN IP4 203.208.207.218
>>>> s=session
>>>> c=IN IP4 203.208.207.195
>>>> t=0 0
>>>> m=audio 45002 RTP/AVP 18 101
>>>> a=rtpmap:18 G729/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>>
>>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:4467 Audio Codec Compare
>>>> [G729:18:8000:20:8000]/[G729:18:8000:20:8000]
>>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:2757 Set Codec
>>>> sofia/external/03996563750914 at 203.208.207.212 G729/8000 20 ms 160
>>>> samples
>>>> 8000 bits
>>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:4565 Set 2833 dtmf send
>>>> payload to 101
>>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:2987 AUDIO RTP
>>>> [sofia/external/03996563750914 at 203.208.207.212] 10.1.1.46 port 28564 ->
>>>> 203.208.207.195 port 45002 codec: 18 ms: 20
>>>> 2011-06-10 11:10:28.513195 [DEBUG] switch_rtp.c:1607 Starting timer
>>>> [soft]
>>>> 160 bytes per 20ms
>>>> 2011-06-10 11:10:28.514938 [DEBUG] sofia_glue.c:3228 Set 2833 dtmf send
>>>> payload to 101
>>>> 2011-06-10 11:10:28.514938 [DEBUG] sofia_glue.c:3233 Set 2833 dtmf
>>>> receive
>>>> payload to 101
>>>> 2011-06-10 11:10:28.514938 [NOTICE] sofia_glue.c:3680 Pre-Answer
>>>> sofia/external/03996563750914 at 203.208.207.212!
>>>> 2011-06-10 11:10:28.514938 [DEBUG] switch_channel.c:2627
>>>> (sofia/external/03996563750914 at 203.208.207.212) Callstate Change
>>>> RINGING ->
>>>> EARLY
>>>> 2011-06-10 11:10:28.514938 [DEBUG] switch_ivr_originate.c:3408 Originate
>>>> Resulted in Success: [sofia/external/03996563750914 at 203.208.207.212]
>>>> 2011-06-10 11:10:28.514938 [DEBUG] mod_commands.c:3205
>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change
>>>> CS_CONSUME_MEDIA -> CS_EXECUTE
>>>> 2011-06-10 11:10:28.514938 [DEBUG] switch_core_session.c:1116 Send
>>>> signal
>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>> 2011-06-10 11:10:28.515884 [DEBUG] switch_core_state_machine.c:320
>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>> CS_EXECUTE
>>>> 2011-06-10 11:10:28.515884 [DEBUG] switch_core_state_machine.c:366
>>>> (sofia/external/03996563750914 at 203.208.207.212) State EXECUTE
>>>> 2011-06-10 11:10:28.515884 [DEBUG] mod_sofia.c:240
>>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA EXECUTE
>>>> 2011-06-10 11:10:28.515884 [DEBUG] switch_core_state_machine.c:157
>>>> sofia/external/03996563750914 at 203.208.207.212 Standard EXECUTE
>>>> EXECUTE sofia/external/03996563750914 at 203.208.207.212 park()
>>>> 2011-06-10 11:10:28.516887 [DEBUG] switch_core_session.c:954 Send signal
>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_ivr_bridge.c:1480
>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_EXECUTE
>>>> ->
>>>> CS_HIBERNATE
>>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:1116 Send
>>>> signal
>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_ivr_bridge.c:1482
>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_EXECUTE
>>>> ->
>>>> CS_HIBERNATE
>>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:1116 Send
>>>> signal
>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:771 Send signal
>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:771 Send signal
>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_ivr_play_say.c:1581 done
>>>> playing
>>>> file
>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_ivr.c:563
>>>> sofia/external/03996590031055 at 203.208.207.212 Command Execute
>>>> playback(tone_stream://%(2000,4000,440,480);loops=10)
>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:366
>>>> (sofia/external/03996563750914 at 203.208.207.212) State EXECUTE going to
>>>> sleep
>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:320
>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>> CS_HIBERNATE
>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:381
>>>> (sofia/external/03996563750914 at 203.208.207.212) State HIBERNATE
>>>> 2011-06-10 11:10:28.519223 [DEBUG] mod_sofia.c:221
>>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA HIBERNATE
>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_ivr_bridge.c:731
>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change
>>>> CS_HIBERNATE ->
>>>> CS_RESET
>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_session.c:1116 Send
>>>> signal
>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:381
>>>> (sofia/external/03996563750914 at 203.208.207.212) State HIBERNATE going
>>>> to
>>>> sleep
>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:320
>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>> CS_RESET
>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:362
>>>> (sofia/external/03996563750914 at 203.208.207.212) State RESET
>>>> 2011-06-10 11:10:28.519223 [DEBUG] mod_sofia.c:165
>>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA RESET
>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_ivr_bridge.c:716
>>>> sofia/external/03996563750914 at 203.208.207.212 CUSTOM RESET
>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:66
>>>> sofia/external/03996563750914 at 203.208.207.212 Standard RESET
>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:362
>>>> (sofia/external/03996563750914 at 203.208.207.212) State RESET going to
>>>> sleep
>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_session.c:709 Send signal
>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>> recv 666 bytes from udp/[203.208.207.212]:5060 at 03:10:30.540814:
>>>>
>>>> ------------------------------------------------------------------------
>>>>   SIP/2.0 183 Session Progress
>>>>   Via: SIP/2.0/UDP
>>>> 10.1.1.46:5080
>>>> ;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>>>   To:
>>>> &lt;sip:03996563750914 at 203.208.207.212
>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>   CSeq: 13501649 INVITE
>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
>>>> SUBSCRIBE, UPDATE
>>>>   Content-Type: application/sdp
>>>>   Content-Length:   186
>>>>
>>>>   v=0
>>>>   o=- 131082 2 IN IP4 203.208.207.218
>>>>   s=session
>>>>   c=IN IP4 203.208.207.195
>>>>   t=0 0
>>>>   m=audio 45002 RTP/AVP 18 101
>>>>   a=rtpmap:18 G729/8000
>>>>   a=rtpmap:101 telephone-event/8000
>>>>   a=sendrecv
>>>>
>>>> ------------------------------------------------------------------------
>>>> 2011-06-10 11:10:30.435309 [INFO] sofia.c:729
>>>> sofia/external/03996563750914 at 203.208.207.212 Update Callee ID to
>>>> "03996563750914" <03996563750914>
>>>> 2011-06-10 11:10:30.435309 [DEBUG] sofia.c:4641 Channel
>>>> sofia/external/03996563750914 at 203.208.207.212 skipping state
>>>> [proceeding][183]
>>>> 2011-06-10 11:10:30.691404 [WARNING] switch_core_session.c:1940 Cannot
>>>> execute app 'playback' media required on an outbound channel that does
>>>> not
>>>> have media established
>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:366
>>>> (sofia/external/03996590031055 at 203.208.207.212) State EXECUTE going to
>>>> sleep
>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>> CS_HIBERNATE
>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:381
>>>> (sofia/external/03996590031055 at 203.208.207.212) State HIBERNATE
>>>> 2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:221
>>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA HIBERNATE
>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:731
>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change
>>>> CS_HIBERNATE ->
>>>> CS_RESET
>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_session.c:1116 Send
>>>> signal
>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:381
>>>> (sofia/external/03996590031055 at 203.208.207.212) State HIBERNATE going
>>>> to
>>>> sleep
>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>> CS_RESET
>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:362
>>>> (sofia/external/03996590031055 at 203.208.207.212) State RESET
>>>> 2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:165
>>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA RESET
>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:716
>>>> sofia/external/03996590031055 at 203.208.207.212 CUSTOM RESET
>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:723
>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_RESET
>>>> ->
>>>> CS_SOFT_EXECUTE
>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_session.c:1116 Send
>>>> signal
>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:362
>>>> (sofia/external/03996590031055 at 203.208.207.212) State RESET going to
>>>> sleep
>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>> CS_SOFT_EXECUTE
>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:372
>>>> (sofia/external/03996590031055 at 203.208.207.212) State SOFT_EXECUTE
>>>> 2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:558 SOFIA SOFT_EXECUTE
>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:741
>>>> sofia/external/03996590031055 at 203.208.207.212 CUSTOM SOFT_EXECUTE
>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:761
>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_RESET
>>>> ->
>>>> CS_SOFT_EXECUTE
>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_session.c:1116 Send
>>>> signal
>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>> CS_SOFT_EXECUTE
>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:372
>>>> (sofia/external/03996563750914 at 203.208.207.212) State SOFT_EXECUTE
>>>> 2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:558 SOFIA SOFT_EXECUTE
>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:741
>>>> sofia/external/03996563750914 at 203.208.207.212 CUSTOM SOFT_EXECUTE
>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:204
>>>> sofia/external/03996563750914 at 203.208.207.212 Standard SOFT_EXECUTE
>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:372
>>>> (sofia/external/03996563750914 at 203.208.207.212) State SOFT_EXECUTE
>>>> going to
>>>> sleep
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2535
>>>> (sofia/external/03996563750914 at 203.208.207.212) Callstate Change EARLY
>>>> ->
>>>> HANGUP
>>>> 2011-06-10 11:10:30.711798 [NOTICE] switch_ivr_originate.c:1045 Hangup
>>>> sofia/external/03996563750914 at 203.208.207.212 [CS_SOFT_EXECUTE]
>>>> [DESTINATION_OUT_OF_ORDER]
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2551 Send signal
>>>> sofia/external/03996563750914 at 203.208.207.212 [KILL]
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send
>>>> signal
>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>> CS_HANGUP
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2535
>>>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change ACTIVE
>>>> ->
>>>> HANGUP
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
>>>> (sofia/external/03996563750914 at 203.208.207.212) State HANGUP
>>>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:457 Channel
>>>> sofia/external/03996563750914 at 203.208.207.212 hanging up, cause:
>>>> DESTINATION_OUT_OF_ORDER
>>>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:510 Sending CANCEL to
>>>> sofia/external/03996563750914 at 203.208.207.212
>>>> send 390 bytes to udp/[203.208.207.212]:5060 at 03:10:30.818391:
>>>>
>>>> ------------------------------------------------------------------------
>>>>   CANCEL sip:03996563750914 at 203.208.207.212 SIP/2.0
>>>>   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKZ49rj5Qtr8aBN
>>>>   Max-Forwards: 70
>>>>   From: "" &lt;sip:03996590031055 at 202.73.56.46&gt;;tag=jDSe6ycN62Zar
>>>>   To: &lt;sip:03996563750914 at 203.208.207.212&gt;
>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>   CSeq: 13501649 CANCEL
>>>>   Reason: Q.850;cause=27;text="DESTINATION_OUT_OF_ORDER"
>>>>   Content-Length: 0
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>> 2011-06-10 11:10:30.711798 [NOTICE] switch_ivr_bridge.c:772 Hangup
>>>> sofia/external/03996590031055 at 203.208.207.212 [CS_SOFT_EXECUTE]
>>>> [ORIGINATOR_CANCEL]
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:46
>>>> sofia/external/03996563750914 at 203.208.207.212 Standard HANGUP, cause:
>>>> DESTINATION_OUT_OF_ORDER
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
>>>> (sofia/external/03996563750914 at 203.208.207.212) State HANGUP going to
>>>> sleep
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:351
>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_HANGUP
>>>> ->
>>>> CS_REPORTING
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send
>>>> signal
>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>> CS_REPORTING
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:617
>>>> (sofia/external/03996563750914 at 203.208.207.212) State REPORTING
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:53
>>>> sofia/external/03996563750914 at 203.208.207.212 Standard REPORTING,
>>>> cause:
>>>> DESTINATION_OUT_OF_ORDER
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:617
>>>> (sofia/external/03996563750914 at 203.208.207.212) State REPORTING going
>>>> to
>>>> sleep
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2551 Send signal
>>>> sofia/external/03996590031055 at 203.208.207.212 [KILL]
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send
>>>> signal
>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:372
>>>> (sofia/external/03996590031055 at 203.208.207.212) State SOFT_EXECUTE
>>>> going to
>>>> sleep
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>> CS_HANGUP
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:345
>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change
>>>> CS_REPORTING ->
>>>> CS_DESTROY
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send
>>>> signal
>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1288 Session 40
>>>> (sofia/external/03996563750914 at 203.208.207.212) Locked, Waiting on
>>>> external
>>>> entities
>>>> 2011-06-10 11:10:30.711798 [NOTICE] switch_core_session.c:1306 Session
>>>> 40
>>>> (sofia/external/03996563750914 at 203.208.207.212) Ended
>>>> 2011-06-10 11:10:30.711798 [NOTICE] switch_core_session.c:1308 Close
>>>> Channel
>>>> sofia/external/03996563750914 at 203.208.207.212 [CS_DESTROY]
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:449
>>>> (sofia/external/03996563750914 at 203.208.207.212) Callstate Change HANGUP
>>>> ->
>>>> DOWN
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:452
>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>> CS_DESTROY
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:462
>>>> (sofia/external/03996563750914 at 203.208.207.212) State DESTROY
>>>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:362
>>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA DESTROY
>>>> 2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
>>>> 0x2aaaac013028 (nil)
>>>> 2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
>>>> 0x2aaaac013028 (nil)
>>>> 2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
>>>> 0x2aaaac013088 (nil)
>>>> 2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
>>>> 0x2aaaac013088 (nil)
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:60
>>>> sofia/external/03996563750914 at 203.208.207.212 Standard DESTROY
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:462
>>>> (sofia/external/03996563750914 at 203.208.207.212) State DESTROY going to
>>>> sleep
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
>>>> (sofia/external/03996590031055 at 203.208.207.212) State HANGUP
>>>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:457 Channel
>>>> sofia/external/03996590031055 at 203.208.207.212 hanging up, cause:
>>>> ORIGINATOR_CANCEL
>>>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:500 Sending BYE to
>>>> sofia/external/03996590031055 at 203.208.207.212
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:46
>>>> sofia/external/03996590031055 at 203.208.207.212 Standard HANGUP, cause:
>>>> ORIGINATOR_CANCEL
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
>>>> (sofia/external/03996590031055 at 203.208.207.212) State HANGUP going to
>>>> sleep
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:351
>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_HANGUP
>>>> ->
>>>> CS_REPORTING
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send
>>>> signal
>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>> CS_REPORTING
>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:617
>>>> (sofia/external/03996590031055 at 203.208.207.212) State REPORTING
>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:53
>>>> sofia/external/03996590031055 at 203.208.207.212 Standard REPORTING,
>>>> cause:
>>>> ORIGINATOR_CANCEL
>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:617
>>>> (sofia/external/03996590031055 at 203.208.207.212) State REPORTING going
>>>> to
>>>> sleep
>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:345
>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change
>>>> CS_REPORTING ->
>>>> CS_DESTROY
>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_session.c:1116 Send
>>>> signal
>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_session.c:1288 Session 39
>>>> (sofia/external/03996590031055 at 203.208.207.212) Locked, Waiting on
>>>> external
>>>> entities
>>>> 2011-06-10 11:10:30.714001 [NOTICE] switch_core_session.c:1306 Session
>>>> 39
>>>> (sofia/external/03996590031055 at 203.208.207.212) Ended
>>>> 2011-06-10 11:10:30.714001 [NOTICE] switch_core_session.c:1308 Close
>>>> Channel
>>>> sofia/external/03996590031055 at 203.208.207.212 [CS_DESTROY]
>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:449
>>>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change HANGUP
>>>> ->
>>>> DOWN
>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:452
>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>> CS_DESTROY
>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:462
>>>> (sofia/external/03996590031055 at 203.208.207.212) State DESTROY
>>>> 2011-06-10 11:10:30.714001 [DEBUG] mod_sofia.c:362
>>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA DESTROY
>>>> 2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
>>>> 0x2aaab0031060 (nil)
>>>> 2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
>>>> 0x2aaab0031060 (nil)
>>>> 2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
>>>> 0x2aaab00310c0 0x2aaab00b20c0
>>>> 2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
>>>> 0x2aaab00310c0 (nil)
>>>> send 662 bytes to udp/[203.208.207.212]:5060 at 03:10:30.820530:
>>>>
>>>> ------------------------------------------------------------------------
>>>>   BYE sip:203.208.207.212:5060 SIP/2.0
>>>>   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bK0D3Hm08XNH1Xg
>>>>   Max-Forwards: 70
>>>>   From: "" &lt;sip:03996563750914 at 202.73.56.46&gt;;tag=H4ZN43UH9S9Qc
>>>>   To:
>>>> &lt;sip:03996590031055 at 203.208.207.212
>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>   CSeq: 13501634 BYE
>>>>   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
>>>>   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>> REGISTER, REFER, NOTIFY
>>>>   Supported: timer, precondition, path, replaces
>>>>   Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL"
>>>>   Content-Length: 0
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>> 2011-06-10 11:10:30.715878 [INFO] mod_com_g729.c:83 ENCODER DESTROY -
>>>> 0x2aaab00310c0 0x2aaab00b20c0
>>>> 2011-06-10 11:10:30.715878 [DEBUG] switch_core_state_machine.c:60
>>>> sofia/external/03996590031055 at 203.208.207.212 Standard DESTROY
>>>> 2011-06-10 11:10:30.715878 [DEBUG] switch_core_state_machine.c:462
>>>> (sofia/external/03996590031055 at 203.208.207.212) State DESTROY going to
>>>> sleep
>>>> recv 383 bytes from udp/[203.208.207.212]:5060 at 03:10:30.823302:
>>>>
>>>> ------------------------------------------------------------------------
>>>>   SIP/2.0 200 Ok
>>>>   Via: SIP/2.0/UDP
>>>> 10.1.1.46:5080
>>>> ;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>>>   To:
>>>> &lt;sip:03996563750914 at 203.208.207.212
>>>> &gt;;tag=2QGB951HCR30000E1D00000u00000001QXU3LU
>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>   CSeq: 13501649 CANCEL
>>>>   Contact: <sip:203.208.207.212:5060>
>>>>   Content-Length:     0
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>> recv 411 bytes from udp/[203.208.207.212]:5060 at 03:10:30.824765:
>>>>
>>>> ------------------------------------------------------------------------
>>>>   SIP/2.0 487 Request Terminated
>>>>   Via: SIP/2.0/UDP
>>>> 10.1.1.46:5080
>>>> ;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>>>   To:
>>>> &lt;sip:03996563750914 at 203.208.207.212
>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>   CSeq: 13501649 INVITE
>>>>   Reason: SIP;cause=487;text="Request Terminated"
>>>>   Content-Length:     0
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>> send 371 bytes to udp/[203.208.207.212]:5060 at 03:10:30.824891:
>>>>
>>>> ------------------------------------------------------------------------
>>>>   ACK sip:03996563750914 at 203.208.207.212 SIP/2.0
>>>>   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKZ49rj5Qtr8aBN
>>>>   Max-Forwards: 70
>>>>   From: "" &lt;sip:03996590031055 at 202.73.56.46&gt;;tag=jDSe6ycN62Zar
>>>>   To:
>>>> &lt;sip:03996563750914 at 203.208.207.212
>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>   CSeq: 13501649 ACK
>>>>   Content-Length: 0
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>> recv 380 bytes from udp/[203.208.207.212]:5060 at 03:10:30.826375:
>>>>
>>>> ------------------------------------------------------------------------
>>>>   SIP/2.0 200 Ok
>>>>   Via: SIP/2.0/UDP
>>>> 10.1.1.46:5080
>>>> ;rport=5080;branch=z9hG4bK0D3Hm08XNH1Xg;received=10.1.1.46
>>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>>   To:
>>>> &lt;sip:03996590031055 at 203.208.207.212
>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>   CSeq: 13501634 BYE
>>>>
>>>>
>>>>
>>>> --
>>>> View this message in context:
>>>> http://freeswitch-users.2379917.n2.nabble.com/Leg-A-is-automatically-disconnected-on-Leg-B-orginate-failure-tp6482192p6482192.html
>>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri Jun 24 10:48:48 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 23 Jun 2011 23:48:48 -0700
Subject: [Freeswitch-users] FreeSWITCH-users@lists.freeswitch.org
In-Reply-To: <1308751418.2780.11.camel@DELL>
References: <1308751418.2780.11.camel@DELL>
Message-ID: <BANLkTimUx5O1Je+x7Y+46ytVPkPWk+LUhQ@mail.gmail.com>

I'm seeing quite a few:


   - [Freeswitch-users] One way audio with Localphone .com
   <http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073570.html>
     *Henk Oegema *
   - [Freeswitch-users] One way audio with Localphone .com
   <http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073584.html>
     *Henk Oegema *
   - [Freeswitch-users] Warning - SIP auth challenge
   <http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073587.html>
     *Henk Oegema *
   - [Freeswitch-users] [Fwd: Re: Warning - SIP auth challenge]
   <http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073589.html>
     *Henk Oegema *
   - [Freeswitch-users] [Fwd: Asterisk as gateway for FS]
   <http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073607.html>
     *Henk Oegema *
   - [Freeswitch-users] No more sessions allowed at this time.
   <http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073768.html>
     *Henk Oegema *
   - [Freeswitch-users] Call internal, otherwise mobile, not working
   <http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073814.html>
     *Henk Oegema *
   - [Freeswitch-users] Call internal, otherwise mobile, not working
   <http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073818.html>
     *Henk Oegema *
   - [Freeswitch-users] FreeSWITCH-users at lists.freeswitch.org
   <http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073848.html>
     *Henk Oegema *

-MC

On Wed, Jun 22, 2011 at 7:03 AM, Henk Oegema <henk at oegema.com> wrote:

> **
> How come I don't see my mails in the FreeSWITCH-Users Archives?
>
>
> Rgds.
> Henk
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri Jun 24 10:52:43 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 23 Jun 2011 23:52:43 -0700
Subject: [Freeswitch-users] Hooks for own call logs
In-Reply-To: <4DFC96BA.1080306@amooma.de>
References: <BANLkTinz0cptgEt3JQ8Eq2FDuNbgsfDdLw@mail.gmail.com>
	<BANLkTim3mx89_p2+KmXk0-iLaBAXQC1EGA@mail.gmail.com>
	<4DFC96BA.1080306@amooma.de>
Message-ID: <BANLkTi=RJMmQx7fk1cOXqF_c_RK8MdCo5A@mail.gmail.com>

Sorry for the late reply...

Have you tried setting the api_hangup_hook variable?
http://wiki.freeswitch.org/wiki/Channel_Variables#api_hangup_hook
-MC

On Sat, Jun 18, 2011 at 5:14 AM, Sascha Daniels <sascha.daniels at amooma.de>wrote:

> Hi.
>
> Am 17.06.2011 03:43, schrieb Michael Collins:
> > It depends on what you mean by "canceled" - can you pastebin a call
> > log of a canceled call and give us an idea what should be happening?
> >
>
> I would like to make an http call to my application after the originator
> canceled the call.
>
> Here ist the cli output.
>
> http://pastebin.com/fCadQTMQ
>
> Somewhere after line 96 the script (or what ever) should be called.
>
> Thanks a lot.
>
> Regards
>
>                    Sascha
>
>
> --
> AMOOMA GmbH - Bachstr. 124 - 56566 Neuwied  -->  http://www.amooma.de
> Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister Montabaur B14998
>
> B?cher: http://das-asterisk-buch.de - http://ruby-auf-schienen.de
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From freeswitch-list at puzzled.xs4all.nl  Fri Jun 24 11:05:37 2011
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Fri, 24 Jun 2011 09:05:37 +0200
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <32B9468B-3402-4056-A18D-D4A01626A87A@gmail.com>
References: <CA293E4C.18131%krice@freeswitch.org>
	<32B9468B-3402-4056-A18D-D4A01626A87A@gmail.com>
Message-ID: <4E043741.8090204@puzzled.xs4all.nl>

On 06/24/2011 03:53 AM, Robert-iPhone wrote:
> isnt voip great - nobody learns telecom 101.
> LERG
> LRN
> LNP
> OCN
> B8ZS / SF

At the risk of being wrong (it's been more than a decade since I worked 
at Lucent), iirc LERG, LRN and OCN are US centric. Since I'm on the 
other side of the Big Blue Pond I don't need to know them :) It probably 
helps if one knows LNP if one offers or uses VoIP services.

B8ZS line encoding. Iirc a funky version of AMI because the US suffered 
from not-invented-here syndrom so they came up with a tweaked version of 
AMI called Bipolar-8-0-substition. Nuf said :) Brings back memories of 
long nights troubleshooting international circuits with BERT testers and 
hopping on planes to follow the nodes so we could finally pinpoint which 
carrier was lying about nothing being wrong with their stretch.

> just the good ole days 5ESS / DMS RCV

5ESS wasn't my thing. They were just really big and did not do very 
exciting things so best left alone with the ancient bearded tribe that 
ran those monsters. All I can remember about the Nortel stuff was that 
it was really loud, particularly their X.25 nodes. I do have fond 
memories of all the Ascend gear (the TNT, CBS-500 and GX-550) and all 
the Lucent experimental DWDM and SDH gear that we got to play with.

"If we build it they will come" - wonder if we will ever see those days 
again.

Regards,
Patrick


From kris at livecall.com  Fri Jun 24 11:25:01 2011
From: kris at livecall.com (Kris)
Date: Fri, 24 Jun 2011 00:25:01 -0700
Subject: [Freeswitch-users] Any RECORD_STOP event sample available?
References: <BANLkTimkbxVe3GxFfaHY-+LxBs5_umpGSA@mail.gmail.com>
Message-ID: <EDD3B1990F6D48E0B2795B4E155083D9@stor1>

Here are the start and stop events.

Hope it helps.



Kris





2,BD16C590-D3E5-4CC8-B551-5E6A7242C38F, 2011/06/24 00:15:47.921875, [INFO], 
Thread Run 2,END 
'E:\LiveMatch\Sounds\en\us\female\SEND_MSG_TO_THIS_PERSON.wav' Digits:1

2,BD16C590-D3E5-4CC8-B551-5E6A7242C38F, 2011/06/24 00:15:48.437500, [INFO], 
Thread Run 2,vAddFile play_info[0] File: 
E:\LiveMatch\Sounds\en\us\female\LEAVE_A_MESSAGE_FOR.wav.

2,BD16C590-D3E5-4CC8-B551-5E6A7242C38F, 2011/06/24 00:15:48.437500, [INFO], 
Thread Run 2,vAddFile play_info[1] File: 
E:\Tele\A\222\202\6\9\2222026917.2222026917.11.96.vox.

2,BD16C590-D3E5-4CC8-B551-5E6A7242C38F, 2011/06/24 00:15:48.468750, [INFO], 
Thread Run 2,BEG read(min_d 0 max_d 1 StartIndx:0 LastIndx:1 max_t0 term 
none intd_t 0)

2,BD16C590-D3E5-4CC8-B551-5E6A7242C38F, 2011/06/24 00:15:51.828125, [INFO], 
Thread Run 2,END 
'E:\LiveMatch\Sounds\en\us\female\LEAVE_A_MESSAGE_FOR.wav&E:\Tele\A\222\202\6\9\2222026917.2222026917.11.96.vox' 
Digits:

2,BD16C590-D3E5-4CC8-B551-5E6A7242C38F, 2011/06/24 00:15:52.921875, [INFO], 
Thread Run 2,BEG 
RecordFile(\\DINGO\E_Drive\tele\M\222\202\6\9\2222026917.2222014124.114.vox,tmoout 
900000,sil_thrsh 1000,sil_hits 10)

0, NO UUID, 2011/06/24 00:15:53.000000, [FSEVENT], Thread 
LiveMatchMaintenanceEvents,<event>

<headers>

<Event-Name>RECORD_START</Event-Name>

<Core-UUID>fdcec719-e099-4919-a33e-71b75d0f5133</Core-UUID>

<FreeSWITCH-Hostname>dingo</FreeSWITCH-Hostname>

<FreeSWITCH-Switchname>dingo</FreeSWITCH-Switchname>

<FreeSWITCH-IPv4>192.168.1.14</FreeSWITCH-IPv4>

<FreeSWITCH-IPv6>%3A%3A1</FreeSWITCH-IPv6>

<Event-Date-Local>2011-06-24%2000%3A15%3A52</Event-Date-Local>

<Event-Date-GMT>Fri,%2024%20Jun%202011%2007%3A15%3A52%20GMT</Event-Date-GMT>

<Event-Date-Timestamp>1308899752937500</Event-Date-Timestamp>

<Event-Calling-File>switch_ivr_play_say.c</Event-Calling-File>

<Event-Calling-Function>switch_ivr_record_file</Event-Calling-Function>

<Event-Calling-Line-Number>615</Event-Calling-Line-Number>

<Channel-State>CS_EXECUTE</Channel-State>

<Channel-Call-State>ACTIVE</Channel-Call-State>

<Channel-State-Number>4</Channel-State-Number>

<Channel-Name>sofia/internal/2014124%40192.168.1.14</Channel-Name>

<Unique-ID>bd16c590-d3e5-4cc8-b551-5e6a7242c38f</Unique-ID>

<Call-Direction>inbound</Call-Direction>

<Presence-Call-Direction>inbound</Presence-Call-Direction>

<Channel-Presence-ID>2014124%40192.168.1.14</Channel-Presence-ID>

<Answer-State>answered</Answer-State>

<Channel-Read-Codec-Name>L16</Channel-Read-Codec-Name>

<Channel-Read-Codec-Rate>8000</Channel-Read-Codec-Rate>

<Channel-Read-Codec-Bit-Rate>128000</Channel-Read-Codec-Bit-Rate>

<Channel-Write-Codec-Name>PCMU</Channel-Write-Codec-Name>

<Channel-Write-Codec-Rate>8000</Channel-Write-Codec-Rate>

<Channel-Write-Codec-Bit-Rate>64000</Channel-Write-Codec-Bit-Rate>

<Caller-Direction>inbound</Caller-Direction>

<Caller-Username>2014124</Caller-Username>

<Caller-Dialplan>XML</Caller-Dialplan>

<Caller-Caller-ID-Name>LaptopXL</Caller-Caller-ID-Name>

<Caller-Caller-ID-Number>2014124</Caller-Caller-ID-Number>

<Caller-Network-Addr>192.168.1.9</Caller-Network-Addr>

<Caller-ANI>2014124</Caller-ANI>

<Caller-Destination-Number>livematch</Caller-Destination-Number>

<Caller-Unique-ID>bd16c590-d3e5-4cc8-b551-5e6a7242c38f</Caller-Unique-ID>

<Caller-Source>mod_sofia</Caller-Source>

<Caller-Context>default</Caller-Context>

<Caller-Channel-Name>sofia/internal/2014124%40192.168.1.14</Caller-Channel-Name>

<Caller-Profile-Index>1</Caller-Profile-Index>

<Caller-Profile-Created-Time>1308899717078125</Caller-Profile-Created-Time>

<Caller-Channel-Created-Time>1308899717078125</Caller-Channel-Created-Time>

<Caller-Channel-Answered-Time>1308899717125000</Caller-Channel-Answered-Time>

<Caller-Channel-Progress-Time>0</Caller-Channel-Progress-Time>

<Caller-Channel-Progress-Media-Time>0</Caller-Channel-Progress-Media-Time>

<Caller-Channel-Hangup-Time>0</Caller-Channel-Hangup-Time>

<Caller-Channel-Transfer-Time>0</Caller-Channel-Transfer-Time>

<Caller-Screen-Bit>true</Caller-Screen-Bit>

<Caller-Privacy-Hide-Name>false</Caller-Privacy-Hide-Name>

<Caller-Privacy-Hide-Number>false</Caller-Privacy-Hide-Number>

<variable_direction>inbound</variable_direction>

<variable_uuid>bd16c590-d3e5-4cc8-b551-5e6a7242c38f</variable_uuid>

<variable_sip_local_network_addr>192.168.1.14</variable_sip_local_network_addr>

<variable_sip_network_ip>192.168.1.9</variable_sip_network_ip>

<variable_sip_network_port>59765</variable_sip_network_port>

<variable_sip_received_ip>192.168.1.9</variable_sip_received_ip>

<variable_sip_received_port>59765</variable_sip_received_port>

<variable_sip_via_protocol>udp</variable_sip_via_protocol>

<variable_sip_authorized>true</variable_sip_authorized>

<variable_sip_number_alias>2014124</variable_sip_number_alias>

<variable_sip_auth_username>2014124</variable_sip_auth_username>

<variable_sip_auth_realm>192.168.1.14</variable_sip_auth_realm>

<variable_number_alias>2014124</variable_number_alias>

<variable_user_name>2014124</variable_user_name>

<variable_domain_name>192.168.1.14</variable_domain_name>

<variable_user_context>default</variable_user_context>

<variable_accountcode>2014124</variable_accountcode>

<variable_toll_allow>local</variable_toll_allow>

<variable_effective_caller_id_name>2014124</variable_effective_caller_id_name>

<variable_effective_caller_id_number>2014124</variable_effective_caller_id_number>

<variable_outbound_caller_id_name>FreeSWITCH</variable_outbound_caller_id_name>

<variable_outbound_caller_id_number>0000000000</variable_outbound_caller_id_number>

<variable_sip_from_user>2014124</variable_sip_from_user>

<variable_sip_from_uri>2014124%40192.168.1.14</variable_sip_from_uri>

<variable_sip_from_host>192.168.1.14</variable_sip_from_host>

<variable_sip_from_user_stripped>2014124</variable_sip_from_user_stripped>

<variable_sofia_profile_name>internal</variable_sofia_profile_name>

<variable_sip_full_via>SIP/2.0/UDP%20192.168.1.9%3A59765%3Bbranch%3Dz9hG4bK-d8754z-4bde784ddbe2a67b-1---d8754z-%3Brport%3D59765</variable_sip_full_via>

<variable_sip_req_user>livematch</variable_sip_req_user>

<variable_sip_req_uri>livematch%40192.168.1.14</variable_sip_req_uri>

<variable_sip_req_host>192.168.1.14</variable_sip_req_host>

<variable_sip_to_user>livematch</variable_sip_to_user>

<variable_sip_to_uri>livematch%40192.168.1.14</variable_sip_to_uri>

<variable_sip_to_host>192.168.1.14</variable_sip_to_host>

<variable_sip_contact_user>2014124</variable_sip_contact_user>

<variable_sip_contact_port>59765</variable_sip_contact_port>

<variable_sip_contact_uri>2014124%40192.168.1.9%3A59765</variable_sip_contact_uri>

<variable_sip_contact_host>192.168.1.9</variable_sip_contact_host>

<variable_channel_name>sofia/internal/2014124%40192.168.1.14</variable_channel_name>

<variable_sip_user_agent>X-Lite%204%20release%204.0%20stamp%2058832</variable_sip_user_agent>

<variable_sip_via_host>192.168.1.9</variable_sip_via_host>

<variable_sip_via_port>59765</variable_sip_via_port>

<variable_sip_via_rport>59765</variable_sip_via_rport>

<variable_max_forwards>70</variable_max_forwards>

<variable_presence_id>2014124%40192.168.1.14</variable_presence_id>

<variable_switch_r_sdp>v%3D0%0D%0Ao%3D-%2012953373323111530%201%20IN%20IP4%20192.168.1.9%0D%0As%3DCounterPath%20X-Lite%204.0%0D%0Ac%3DIN%20IP4%20192.168.1.9%0D%0At%3D0%200%0D%0Am%3Daudio%2064060%20RTP/AVP%20107%200%208%203%20101%0D%0Aa%3Drtpmap%3A107%20BV32/16000%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A</variable_switch_r_sdp>

<variable_sip_audio_recv_pt>0</variable_sip_audio_recv_pt>

<variable_sip_use_codec_name>PCMU</variable_sip_use_codec_name>

<variable_sip_use_codec_rate>8000</variable_sip_use_codec_rate>

<variable_sip_use_codec_ptime>20</variable_sip_use_codec_ptime>

<variable_write_codec>PCMU</variable_write_codec>

<variable_write_rate>8000</variable_write_rate>

<variable_tts_engine>flite</variable_tts_engine>

<variable_tts_voice>kal</variable_tts_voice>

<variable_current_application_data>LiveMatchApp%20web_conference_menu</variable_current_application_data>

<variable_current_application>managed</variable_current_application>

<variable_LM_CallID>2</variable_LM_CallID>

<variable_sip_local_sdp_str>v%3D0%0Ao%3DFreeSWITCH%201308874795%201308874796%20IN%20IP4%20192.168.1.14%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%20192.168.1.14%0At%3D0%200%0Am%3Daudio%2024922%20RTP/AVP%200%20101%0Aa%3Drtpmap%3A0%20PCMU/8000%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A</variable_sip_local_sdp_str>

<variable_local_media_ip>192.168.1.14</variable_local_media_ip>

<variable_local_media_port>24922</variable_local_media_port>

<variable_sip_use_pt>0</variable_sip_use_pt>

<variable_rtp_use_ssrc>1294177623</variable_rtp_use_ssrc>

<variable_remote_media_ip>192.168.1.9</variable_remote_media_ip>

<variable_remote_media_port>64060</variable_remote_media_port>

<variable_endpoint_disposition>ANSWER</variable_endpoint_disposition>

<variable_sip_to_tag>918XvDZ82QZFK</variable_sip_to_tag>

<variable_sip_from_tag>bf5785fd</variable_sip_from_tag>

<variable_sip_cseq>3</variable_sip_cseq>

<variable_sip_call_id>MmM0OWIzNWNkYmY4MWZhMTk4NTQ1Zjg0NTU5NGIzMTU.</variable_sip_call_id>

<variable_sip_from_display>LaptopXL</variable_sip_from_display>

<variable_sip_full_from>%22LaptopXL%22%20%3Csip%3A2014124%40192.168.1.14%3E%3Btag%3Dbf5785fd</variable_sip_full_from>

<variable_sip_to_display>livematch</variable_sip_to_display>

<variable_sip_full_to>%22livematch%22%20%3Csip%3Alivematch%40192.168.1.14%3E%3Btag%3D918XvDZ82QZFK</variable_sip_full_to>

<variable_utc_start_time>06/24/2011%2007%3A15%3A17.843</variable_utc_start_time>

<variable_CallerExt>2222014124</variable_CallerExt>

<variable_cdr_csv_base>%5C%5CDINGO%5CE_Drive%5Ctele%5CCDR%5C222%5C201%5C4%5C1%5C</variable_cdr_csv_base>

<variable_playback_delimiter>%26</variable_playback_delimiter>

<variable_playback_sleep_val>250</variable_playback_sleep_val>

<variable_read_result>failure</variable_read_result>

<variable_record_sample_rate>8000</variable_record_sample_rate>

<variable_playback_seconds>0</variable_playback_seconds>

<variable_playback_ms>900</variable_playback_ms>

<variable_playback_samples>7200</variable_playback_samples>

<variable_read_codec>L16</variable_read_codec>

<variable_read_rate>8000</variable_read_rate>

<Record-File-Path>%5C%5CDINGO%5CE_Drive%5Ctele%5CM%5C222%5C202%5C6%5C9%5C2222026917.2222014124.114.vox</Record-File-Path>

</headers>

</event>

2,bd16c590-d3e5-4cc8-b551-5e6a7242c38f, 2011/06/24 00:15:53.031250, [INFO], 
Thread LiveMatchMaintenanceEvents,Started Record .

2,BD16C590-D3E5-4CC8-B551-5E6A7242C38F, 2011/06/24 00:16:02.921875, [INFO], 
Thread Run 2,END 
RecordFile(\\DINGO\E_Drive\tele\M\222\202\6\9\2222026917.2222014124.114.vox)result:0 
Digits:

0, NO UUID, 2011/06/24 00:16:03.000000, [FSEVENT], Thread 
LiveMatchMaintenanceEvents,<event>

<headers>

<Event-Name>RECORD_STOP</Event-Name>

<Core-UUID>fdcec719-e099-4919-a33e-71b75d0f5133</Core-UUID>

<FreeSWITCH-Hostname>dingo</FreeSWITCH-Hostname>

<FreeSWITCH-Switchname>dingo</FreeSWITCH-Switchname>

<FreeSWITCH-IPv4>192.168.1.14</FreeSWITCH-IPv4>

<FreeSWITCH-IPv6>%3A%3A1</FreeSWITCH-IPv6>

<Event-Date-Local>2011-06-24%2000%3A16%3A02</Event-Date-Local>

<Event-Date-GMT>Fri,%2024%20Jun%202011%2007%3A16%3A02%20GMT</Event-Date-GMT>

<Event-Date-Timestamp>1308899762921875</Event-Date-Timestamp>

<Event-Calling-File>switch_ivr_play_say.c</Event-Calling-File>

<Event-Calling-Function>switch_ivr_record_file</Event-Calling-Function>

<Event-Calling-Line-Number>767</Event-Calling-Line-Number>

<Channel-State>CS_EXECUTE</Channel-State>

<Channel-Call-State>ACTIVE</Channel-Call-State>

<Channel-State-Number>4</Channel-State-Number>

<Channel-Name>sofia/internal/2014124%40192.168.1.14</Channel-Name>

<Unique-ID>bd16c590-d3e5-4cc8-b551-5e6a7242c38f</Unique-ID>

<Call-Direction>inbound</Call-Direction>

<Presence-Call-Direction>inbound</Presence-Call-Direction>

<Channel-Presence-ID>2014124%40192.168.1.14</Channel-Presence-ID>

<Answer-State>answered</Answer-State>

<Channel-Read-Codec-Name>L16</Channel-Read-Codec-Name>

<Channel-Read-Codec-Rate>8000</Channel-Read-Codec-Rate>

<Channel-Read-Codec-Bit-Rate>128000</Channel-Read-Codec-Bit-Rate>

<Channel-Write-Codec-Name>PCMU</Channel-Write-Codec-Name>

<Channel-Write-Codec-Rate>8000</Channel-Write-Codec-Rate>

<Channel-Write-Codec-Bit-Rate>64000</Channel-Write-Codec-Bit-Rate>

<Caller-Direction>inbound</Caller-Direction>

<Caller-Username>2014124</Caller-Username>

<Caller-Dialplan>XML</Caller-Dialplan>

<Caller-Caller-ID-Name>LaptopXL</Caller-Caller-ID-Name>

<Caller-Caller-ID-Number>2014124</Caller-Caller-ID-Number>

<Caller-Network-Addr>192.168.1.9</Caller-Network-Addr>

<Caller-ANI>2014124</Caller-ANI>

<Caller-Destination-Number>livematch</Caller-Destination-Number>

<Caller-Unique-ID>bd16c590-d3e5-4cc8-b551-5e6a7242c38f</Caller-Unique-ID>

<Caller-Source>mod_sofia</Caller-Source>

<Caller-Context>default</Caller-Context>

<Caller-Channel-Name>sofia/internal/2014124%40192.168.1.14</Caller-Channel-Name>

<Caller-Profile-Index>1</Caller-Profile-Index>

<Caller-Profile-Created-Time>1308899717078125</Caller-Profile-Created-Time>

<Caller-Channel-Created-Time>1308899717078125</Caller-Channel-Created-Time>

<Caller-Channel-Answered-Time>1308899717125000</Caller-Channel-Answered-Time>

<Caller-Channel-Progress-Time>0</Caller-Channel-Progress-Time>

<Caller-Channel-Progress-Media-Time>0</Caller-Channel-Progress-Media-Time>

<Caller-Channel-Hangup-Time>0</Caller-Channel-Hangup-Time>

<Caller-Channel-Transfer-Time>0</Caller-Channel-Transfer-Time>

<Caller-Screen-Bit>true</Caller-Screen-Bit>

<Caller-Privacy-Hide-Name>false</Caller-Privacy-Hide-Name>

<Caller-Privacy-Hide-Number>false</Caller-Privacy-Hide-Number>

<variable_direction>inbound</variable_direction>

<variable_uuid>bd16c590-d3e5-4cc8-b551-5e6a7242c38f</variable_uuid>

<variable_sip_local_network_addr>192.168.1.14</variable_sip_local_network_addr>

<variable_sip_network_ip>192.168.1.9</variable_sip_network_ip>

<variable_sip_network_port>59765</variable_sip_network_port>

<variable_sip_received_ip>192.168.1.9</variable_sip_received_ip>

<variable_sip_received_port>59765</variable_sip_received_port>

<variable_sip_via_protocol>udp</variable_sip_via_protocol>

<variable_sip_authorized>true</variable_sip_authorized>

<variable_sip_number_alias>2014124</variable_sip_number_alias>

<variable_sip_auth_username>2014124</variable_sip_auth_username>

<variable_sip_auth_realm>192.168.1.14</variable_sip_auth_realm>

<variable_number_alias>2014124</variable_number_alias>

<variable_user_name>2014124</variable_user_name>

<variable_domain_name>192.168.1.14</variable_domain_name>

<variable_user_context>default</variable_user_context>

<variable_accountcode>2014124</variable_accountcode>

<variable_toll_allow>local</variable_toll_allow>

<variable_effective_caller_id_name>2014124</variable_effective_caller_id_name>

<variable_effective_caller_id_number>2014124</variable_effective_caller_id_number>

<variable_outbound_caller_id_name>FreeSWITCH</variable_outbound_caller_id_name>

<variable_outbound_caller_id_number>0000000000</variable_outbound_caller_id_number>

<variable_sip_from_user>2014124</variable_sip_from_user>

<variable_sip_from_uri>2014124%40192.168.1.14</variable_sip_from_uri>

<variable_sip_from_host>192.168.1.14</variable_sip_from_host>

<variable_sip_from_user_stripped>2014124</variable_sip_from_user_stripped>

<variable_sofia_profile_name>internal</variable_sofia_profile_name>

<variable_sip_full_via>SIP/2.0/UDP%20192.168.1.9%3A59765%3Bbranch%3Dz9hG4bK-d8754z-4bde784ddbe2a67b-1---d8754z-%3Brport%3D59765</variable_sip_full_via>

<variable_sip_req_user>livematch</variable_sip_req_user>

<variable_sip_req_uri>livematch%40192.168.1.14</variable_sip_req_uri>

<variable_sip_req_host>192.168.1.14</variable_sip_req_host>

<variable_sip_to_user>livematch</variable_sip_to_user>

<variable_sip_to_uri>livematch%40192.168.1.14</variable_sip_to_uri>

<variable_sip_to_host>192.168.1.14</variable_sip_to_host>

<variable_sip_contact_user>2014124</variable_sip_contact_user>

<variable_sip_contact_port>59765</variable_sip_contact_port>

<variable_sip_contact_uri>2014124%40192.168.1.9%3A59765</variable_sip_contact_uri>

<variable_sip_contact_host>192.168.1.9</variable_sip_contact_host>

<variable_channel_name>sofia/internal/2014124%40192.168.1.14</variable_channel_name>

<variable_sip_user_agent>X-Lite%204%20release%204.0%20stamp%2058832</variable_sip_user_agent>

<variable_sip_via_host>192.168.1.9</variable_sip_via_host>

<variable_sip_via_port>59765</variable_sip_via_port>

<variable_sip_via_rport>59765</variable_sip_via_rport>

<variable_max_forwards>70</variable_max_forwards>

<variable_presence_id>2014124%40192.168.1.14</variable_presence_id>

<variable_switch_r_sdp>v%3D0%0D%0Ao%3D-%2012953373323111530%201%20IN%20IP4%20192.168.1.9%0D%0As%3DCounterPath%20X-Lite%204.0%0D%0Ac%3DIN%20IP4%20192.168.1.9%0D%0At%3D0%200%0D%0Am%3Daudio%2064060%20RTP/AVP%20107%200%208%203%20101%0D%0Aa%3Drtpmap%3A107%20BV32/16000%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A</variable_switch_r_sdp>

<variable_sip_audio_recv_pt>0</variable_sip_audio_recv_pt>

<variable_sip_use_codec_name>PCMU</variable_sip_use_codec_name>

<variable_sip_use_codec_rate>8000</variable_sip_use_codec_rate>

<variable_sip_use_codec_ptime>20</variable_sip_use_codec_ptime>

<variable_write_codec>PCMU</variable_write_codec>

<variable_write_rate>8000</variable_write_rate>

<variable_tts_engine>flite</variable_tts_engine>

<variable_tts_voice>kal</variable_tts_voice>

<variable_current_application_data>LiveMatchApp%20web_conference_menu</variable_current_application_data>

<variable_current_application>managed</variable_current_application>

<variable_LM_CallID>2</variable_LM_CallID>

<variable_sip_local_sdp_str>v%3D0%0Ao%3DFreeSWITCH%201308874795%201308874796%20IN%20IP4%20192.168.1.14%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%20192.168.1.14%0At%3D0%200%0Am%3Daudio%2024922%20RTP/AVP%200%20101%0Aa%3Drtpmap%3A0%20PCMU/8000%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A</variable_sip_local_sdp_str>

<variable_local_media_ip>192.168.1.14</variable_local_media_ip>

<variable_local_media_port>24922</variable_local_media_port>

<variable_sip_use_pt>0</variable_sip_use_pt>

<variable_rtp_use_ssrc>1294177623</variable_rtp_use_ssrc>

<variable_remote_media_ip>192.168.1.9</variable_remote_media_ip>

<variable_remote_media_port>64060</variable_remote_media_port>

<variable_endpoint_disposition>ANSWER</variable_endpoint_disposition>

<variable_sip_to_tag>918XvDZ82QZFK</variable_sip_to_tag>

<variable_sip_from_tag>bf5785fd</variable_sip_from_tag>

<variable_sip_cseq>3</variable_sip_cseq>

<variable_sip_call_id>MmM0OWIzNWNkYmY4MWZhMTk4NTQ1Zjg0NTU5NGIzMTU.</variable_sip_call_id>

<variable_sip_from_display>LaptopXL</variable_sip_from_display>

<variable_sip_full_from>%22LaptopXL%22%20%3Csip%3A2014124%40192.168.1.14%3E%3Btag%3Dbf5785fd</variable_sip_full_from>

<variable_sip_to_display>livematch</variable_sip_to_display>

<variable_sip_full_to>%22livematch%22%20%3Csip%3Alivematch%40192.168.1.14%3E%3Btag%3D918XvDZ82QZFK</variable_sip_full_to>

<variable_utc_start_time>06/24/2011%2007%3A15%3A17.843</variable_utc_start_time>

<variable_CallerExt>2222014124</variable_CallerExt>

<variable_cdr_csv_base>%5C%5CDINGO%5CE_Drive%5Ctele%5CCDR%5C222%5C201%5C4%5C1%5C</variable_cdr_csv_base>

<variable_playback_delimiter>%26</variable_playback_delimiter>

<variable_playback_sleep_val>250</variable_playback_sleep_val>

<variable_read_result>failure</variable_read_result>

<variable_record_sample_rate>8000</variable_record_sample_rate>

<variable_playback_seconds>0</variable_playback_seconds>

<variable_playback_ms>900</variable_playback_ms>

<variable_playback_samples>7200</variable_playback_samples>

<variable_read_codec>L16</variable_read_codec>

<variable_read_rate>8000</variable_read_rate>

<variable_record_seconds>11</variable_record_seconds>

<variable_record_ms>11448</variable_record_ms>

<variable_record_samples>91584</variable_record_samples>

<Record-File-Path>%5C%5CDINGO%5CE_Drive%5Ctele%5CM%5C222%5C202%5C6%5C9%5C2222026917.2222014124.114.vox</Record-File-Path>

</headers>

</event>

2,bd16c590-d3e5-4cc8-b551-5e6a7242c38f, 2011/06/24 00:16:03.000000, [INFO], 
Thread LiveMatchMaintenanceEvents,Stopped Record .

2,BD16C590-D3E5-4CC8-B551-5E6A7242C38F, 2011/06/24 00:16:03.468750, [INFO], 
Thread Run 2,ttsAddText play_info[0] "To send press 1, to re-play press 2, 
to re-record, press 3, To cancel press 7..".

2,BD16C590-D3E5-4CC8-B551-5E6A7242C38F, 2011/06/24 00:16:03.500000, [INFO], 
Thread Run 2,BEG read(min_d 0 max_d 1 StartIndx:0 LastIndx:0 max_t5000 term 
none intd_t 5000)

0, NO UUID, 2011/06/24 00:16:06.218750, [FSEVENT], Thread 
LiveMatchMaintenanceEvents,<event>

<headers>

<Event-Name>HEARTBEAT</Event-Name>

<Core-UUID>fdcec719-e099-4919-a33e-71b75d0f5133</Core-UUID>

<FreeSWITCH-Hostname>dingo</FreeSWITCH-Hostname>

<FreeSWITCH-Switchname>dingo</FreeSWITCH-Switchname>

<FreeSWITCH-IPv4>192.168.1.14</FreeSWITCH-IPv4>

<FreeSWITCH-IPv6>%3A%3A1</FreeSWITCH-IPv6>

<Event-Date-Local>2011-06-24%2000%3A16%3A06</Event-Date-Local>

<Event-Date-GMT>Fri,%2024%20Jun%202011%2007%3A16%3A06%20GMT</Event-Date-GMT>

<Event-Date-Timestamp>1308899766218750</Event-Date-Timestamp>

<Event-Calling-File>switch_core.c</Event-Calling-File>

<Event-Calling-Function>send_heartbeat</Event-Calling-Function>

<Event-Calling-Line-Number>65</Event-Calling-Line-Number>

<Event-Info>System%20Ready</Event-Info>

<Up-Time>0%20years,%200%20days,%200%20hours,%204%20minutes,%2040%20seconds,%20640%20milliseconds,%20625%20microseconds</Up-Time>

<Session-Count>2</Session-Count>

<Session-Per-Sec>30</Session-Per-Sec>

<Session-Since-Startup>3</Session-Since-Startup>

<Idle-CPU>100.000000</Idle-CPU>

</headers>

</event>

2,BD16C590-D3E5-4CC8-B551-5E6A7242C38F, 2011/06/24 00:16:16.140625, [INFO], 
Thread Run 2,END 'say:flite:kal:To send press 1, to re-play press 2, to 
re-record, press 3, To cancel press 7..' Digits:

2,BD16C590-D3E5-4CC8-B551-5E6A7242C38F, 2011/06/24 00:16:16.265625, [INFO], 
Thread Run 2,New message added. Receiver:2222026917 CallerExt: 2222014124 
CallerANI:



//capturing the events and doing something

case "RECORD_START":

if (LoadingNewDLL)

break;

BaseLog.WriteLine(BaseLogLevel.Info, CallID, SessionUUID, Actions.RECORDING, 
TerminationReasons.NONE, "Started Record .");

break;

case "RECORD_STOP":

if (LoadingNewDLL) //allow anything to be processed in the just loaded DLL 
to avoid conflicts

break;

BaseLog.WriteLine(BaseLogLevel.Info, CallID, SessionUUID, Actions.RECORDING, 
TerminationReasons.END_OF_RECORD, "Stopped Record .");

break;

----- Original Message ----- 
From: "Frankie Yiu" <frankie.k.yiu at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, June 23, 2011 7:35 PM
Subject: [Freeswitch-users] Any RECORD_STOP event sample available?


> Hi there,
>
> Could someone please point me where I can find a sample of the RECORD_STOP
> event look like?
> I look at this page but it does not have any information about it. (
> http://wiki.freeswitch.org/wiki/Event_list#RECORD_STOP)
>
> Thanks,
> Frankie
> 



From mail at jankubr.com  Fri Jun 24 13:55:53 2011
From: mail at jankubr.com (Jan Kubr)
Date: Fri, 24 Jun 2011 11:55:53 +0200
Subject: [Freeswitch-users] Loading Pocketsphinx
In-Reply-To: <3277E95E06DC465F885DD89B0FFB7355@dell9400>
References: <BANLkTimjN9=4DH57c9fk37UKKEu8aZ=emw@mail.gmail.com>
	<3277E95E06DC465F885DD89B0FFB7355@dell9400>
Message-ID: <BANLkTi=KTiiTpqgGR0OXmP0Nti4JRr+5HQ@mail.gmail.com>

This is how I got pocketsphinx working on an Ubuntu box:

Downloaded latest SVN snapshots of sphinxbase and pocketsphinx from
http://cmusphinx.sourceforge.net/wiki/download/ to the libs directory of the
FreeSWITCH source.
Unpacked the files and named the directories pocketsphinx-0.7
and sphinxbase-0.7.
Run ./autogen.sh in sphinxbase-0.7
Run make mod_pocketsphinx in the root directory of the source
Run ./authogen.sh in pocketsphinx-0.7
Run make mod_pocketsphinx in the root directory of the source
cp libs/sphinxbase-0.7/src/libsphinxad/.libs/libsphinxad.so*
/usr/local/freeswitch/lib/
cp libs/sphinxbase-0.7/src/libsphinxbase/.libs/libsphinxbase.so*
/usr/local/freeswitch/lib/
Run make mod_pocketsphinx-install in the root directory of the source

Hope this helps.

Jan Kubr

On Mon, Jun 20, 2011 at 5:22 PM, Jan Berger <jan.berger at video24.no> wrote:

>  You can also use pocketsphinx through uniMRCP according to their website.
> I have not tried it myself yet but I will a bit later.****
>
> ** **
>
> Jan****
>
> ** **
>  ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Bryan Lemon
> *Sent:* 19. juni 2011 22:47
> *To:* freeswitch-users
> *Subject:* [Freeswitch-users] Loading Pocketsphinx****
>
> ** **
>
> Whenever I try to load the pocketsphinx module, either on startup, or with
> the load mod_pocketsphinx command, I get the following error:****
>
> freeswitch at bryanlemon-laptop> load mod_pocketsphinx****
>
> 2011-06-19 15:17:04.471275 [INFO] mod_enum.c:775 ENUM Reloaded****
>
> 2011-06-19 15:17:04.471275 [INFO] switch_time.c:1020 Timezone reloaded 530
> definitions****
>
> ** **
>
> +OK Reloading XML****
>
> -ERR [module load file routine returned an error]****
>
> ** **
>
> 2011-06-19 15:17:04.471275 [CRIT] switch_loadable_module.c:928 Error
> Loading module /usr/local/freeswitch/mod/mod_pocketsphinx.so****
>
> **/usr/local/freeswitch/mod/mod_pocketsphinx.so: undefined symbol:
> feat_init******
>
> ** **
>
> ** **
>
> I am running Ubuntu 10.04, and installed freeswitch using the git method. I
> tried a git pull && make current, and it runs through with no problem. ***
> *
>
> ** **
>
> Any Ideas?
>
> Thank you,
> Bryan Lemon****
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Suneel.Papineni at mettoni.com  Fri Jun 24 15:46:09 2011
From: Suneel.Papineni at mettoni.com (Suneel Papineni)
Date: Fri, 24 Jun 2011 12:46:09 +0100
Subject: [Freeswitch-users] Issue with format value 19 in SDP
Message-ID: <3181A30B8C35AB4AA8577B78DDF461380831F32F@nickel.mettonigroup.com>

Hi,

 

I am facing an issue with SDP. FS is getting an INVITE with media
parameters in SDP as "m=image 23456 udptl t38". For this FS is
responding with "m=image 0 udptl 19" in its 183 response. This is
correct as per RFC's. 

 

Unfortunately my ITSP is expecting the same format what it is sent (i.e.
t38 instead of 19). Is there any way in Freeswitch I can specify to the
send same format values in response. 

 

I tried to look into the code, it is mentioned that 19 is used as it is
not used by anybody it is used. (sdp_print.c in libsofia_sip_ua_static
project).

 

Could someone please help me how to send all format values that were
send in INVITE. Of course according to SDP standards it is enough to
send one value. I want to know how to send at least one value which is
same as sent in INVITE.

 

Thanks & Regards

Suneel

 

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From kris at kriskinc.com  Fri Jun 24 17:57:43 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Fri, 24 Jun 2011 09:57:43 -0400
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <3DF51B5A-A97F-42AE-9C31-90D132B4F066@gmail.com>
References: <CA293E4C.18131%krice@freeswitch.org>
	<32B9468B-3402-4056-A18D-D4A01626A87A@gmail.com>
	<1769702213133480228@unknownmsgid>
	<3DF51B5A-A97F-42AE-9C31-90D132B4F066@gmail.com>
Message-ID: <BANLkTin4TGpX6wp2pCRa3Vvz7NY3eWs-dA@mail.gmail.com>

"voip requires less knowledge"

  These kinds of statements only serve to fracture
telephony/communications communities even more (at a time when
"unified communications" have been the buzzword for a decade).

  I take it you've never spent time developing high scale, high
availability "voip" systems.  Built SS7/SIP internetworking gateways?
SIP-I?  SIP-T?  Debugged ridiculous SIP/SDP/RTP interop issues between
the thousands of endpoints, networks, and configurations available in
the field?  TLS?  SRTP?  IPV6?  Complex codecs operating at multiple
sample rates and ptimes, complete with adaptive jitter buffers and
forward error correction?  I could go on, and on, and on and of course
most of us also have to deal with the 8k TDM PSTN (including SS7,
IMTs, various DBs, regulatory stuff, etc, etc).

  Saying "voip requires less knowledge" does nothing more than show
how little you know about it.  Voip does not require less knowledge -
it requires /different/ knowledge.  At this point I couldn't patch a
25 pair cable if my life depended on it, that requires expertise
different from my own.  If you want to talk about any of the
technologies above I'd probably be able to contribute quite a bit to
the conversation :).

On Thu, Jun 23, 2011 at 10:27 PM, Robert-iPhone <rhuddleston at gmail.com> wrote:
> it wasnt intended as insult - just humorous how voip requires less knowledge
>
> Sent from my iPhone

-- 
Kristian Kielhofner


From bmoore at statirasystems.com  Fri Jun 24 19:24:24 2011
From: bmoore at statirasystems.com (William Moore)
Date: Fri, 24 Jun 2011 11:24:24 -0400
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <CA293246.2B652%d@d-man.org>
References: <CA293246.2B652%d@d-man.org>
Message-ID: <4E04AC28.5050402@statirasystems.com>

LOL I know it is not difficult. The change I had an issue storing was 
the expire seconds. It actually stayed on the fifth try. Not sure why it 
would not but it is there. Actually authentication was not the issue 
after all. Junction sends the number with a one so I had to add it to 
the Number setting. Also I had the context setting on the number wrong. 
It kept trying to send it out the second it came in.

My concerns about the project were not about the activity of the 
project. It was about it's attention to the users. I tried the 
documentation, which is better then a lot of projects I have looked at, 
and it failed me. It is only about each specific thing and yet it is not 
that specific. To someone well versed in freeswitch, they could connect 
the dots, my green self could not.

Also that comments I made were because I had attempted assistance via 
2600hz IRC and forum. As you can see I ended up here.


William J. Moore
Statira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com


On 06/23/2011 09:13 PM, Darren Schreiber wrote:
> Hi William,
> Sorry to hear that ? what exactly makes you think this? We're very 
> active on the blue.box front. There have been 3 new features and over 
> 20 bug fixes in just the past four weeks. Can you email me off-list to 
> get whatever issue you're having sorted out?
>
> Also, for authentication, bluebox "replaces" your changes because 
> you're supposed to use the GUI to change that option. Go into 
> Connectivity / SIP Interfaces and edit the SIP port you're trying to 
> change. Uncheck the mark named "Require Authentication". It's not very 
> difficult?
>
> - Darren
>
> -- 
>
>
> From: William Moore <bmoore at statirasystems.com 
> <mailto:bmoore at statirasystems.com>>
> Reply-To: "freeswitch-users at lists.freeswitch.org 
> <mailto:freeswitch-users at lists.freeswitch.org>" 
> <freeswitch-users at lists.freeswitch.org 
> <mailto:freeswitch-users at lists.freeswitch.org>>
> Date: Thu, 23 Jun 2011 10:55:15 -0700
> To: "freeswitch-users at lists.freeswitch.org 
> <mailto:freeswitch-users at lists.freeswitch.org>" 
> <freeswitch-users at lists.freeswitch.org 
> <mailto:freeswitch-users at lists.freeswitch.org>>
> Subject: Re: [Freeswitch-users] Trunk Issue
>
> I give up on blue.box. They seem preoccupied by there other projects. 
> Documentation does not explain enough. It is like describing a human 
> by their parts only and not how it interacts. Peace out.
> William J. Moore
> Statira Systems
> 611 Caroline St
> Fredericksburg, VA
> http://www.statirasystems.com
>
> On 06/23/2011 01:34 PM, David Ponzone wrote:
>> In sipinterface_2, make the following modification:
>>
>> <param name="auth-calls" value="false"/>
>>
>>
>> David Ponzone Direction Technique
>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service ClientIPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
>> <BLOCKED::http://www.ipeva-studio.com/>
>>
>> /Ce message et toutes les pi?ces jointes sont confidentiels et 
>> ?tablis ? l'intention exclusive de ses destinataires. Toute 
>> utilisation ou diffusion non autoris?e est interdite. Tout message 
>> ?lectronique est susceptible d'alt?ration. /*/IPeva/*/ d?cline toute 
>> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou 
>> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le 
>> d?truire imm?diatement et d'avertir l'exp?diteur./
>> /
>> /
>>
>>
>>
>> Le 23/06/2011 ? 19:31, David Ponzone a ?crit :
>>
>>> The issue is that you are expecting them to authenticate, but they 
>>> won't.
>>>
>>> David Ponzone Direction Technique
>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service ClientIPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
>>> <BLOCKED::http://www.ipeva-studio.com/>
>>>
>>> /Ce message et toutes les pi?ces jointes sont confidentiels et 
>>> ?tablis ? l'intention exclusive de ses destinataires. Toute 
>>> utilisation ou diffusion non autoris?e est interdite. Tout message 
>>> ?lectronique est susceptible d'alt?ration. /*/IPeva/*/ d?cline toute 
>>> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou 
>>> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le 
>>> d?truire imm?diatement et d'avertir l'exp?diteur./
>>> /
>>> /
>>>
>>>
>>>
>>> Le 23/06/2011 ? 19:15, William Moore a ?crit :
>>>
>>>> Ok, I reregistered and called right away. I get the following:
>>>>    0.000000 66.227.100.20 ->  192.168.1.254 SIP/SDP Request: INVITEsip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with session description
>>>>    0.000766 192.168.1.254 ->  66.227.100.20 SIP Status: 100 Trying
>>>>    0.002321 192.168.1.254 ->  66.227.100.20 SIP Status: 407 Proxy Authentication Required
>>>>    0.021120 66.227.100.20 ->  192.168.1.254 SIP Request: ACKsip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>>>    3.272168 192.168.1.254 ->  66.227.100.20 SIP Request: REGISTERsip:jnctn.net;transport=udp
>>>>    3.292076 66.227.100.20 ->  192.168.1.254 SIP Status: 200 OK    (1 bindings)
>>>>
>>>> The call just states it is a none working number. That means something in my dial plan is incorrect. I am double checking it now.
>>>> William J. MooreStatira Systems
>>>> 611 Caroline St
>>>> Fredericksburg, VA
>>>> http://www.statirasystems.com
>>>>
>>>> On 06/23/2011 12:58 PM, William Moore wrote:
>>>>> When I put it in and it reloads, blue.box removes it. Can I 
>>>>> reference the gateway in another sip interface file or does it 
>>>>> have to be in that one?
>>>>> William J. Moore
>>>>> Statira Systems
>>>>> 611 Caroline St
>>>>> Fredericksburg, VA
>>>>> http://www.statirasystems.com
>>>>>
>>>>> On 06/23/2011 12:52 PM, William Moore wrote:
>>>>>> correction: missed a line.
>>>>>>
>>>>>>   0.000000 66.227.100.20 -> 192.168.1.254 SIP/SDP Request: INVITE 
>>>>>> sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with 
>>>>>> session description
>>>>>>   0.000814 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>>>>>>   0.002440 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy 
>>>>>> Authentication Required
>>>>>>   0.023173 66.227.100.20 -> 192.168.1.254 SIP Request: ACK 
>>>>>> sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>>>>>
>>>>>> William J. Moore
>>>>>> Statira Systems
>>>>>> 611 Caroline St
>>>>>> Fredericksburg, VA
>>>>>> http://www.statirasystems.com
>>>>>>
>>>>>> On 06/23/2011 12:50 PM, William Moore wrote:
>>>>>>> I get this with wireshark
>>>>>>>
>>>>>>>  40.345679 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>>>>>>>  40.347381 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy 
>>>>>>> Authentication Required
>>>>>>>  40.367906 66.227.100.20 -> 192.168.1.254 SIP Request: ACK 
>>>>>>> sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>>>>>>
>>>>>>> I am looking for the expiration now..
>>>>>>> William J. Moore
>>>>>>> Statira Systems
>>>>>>> 611 Caroline St
>>>>>>> Fredericksburg, VA
>>>>>>> http://www.statirasystems.com
>>>>>>>
>>>>>>> On 06/23/2011 12:25 PM, David Ponzone wrote:
>>>>>>>> Take a sip trace on your box with tcpdump or preferably, tshark:
>>>>>>>> tshark port 5080
>>>>>>>>
>>>>>>>> You will then see if you receive it.
>>>>>>>>
>>>>>>>> In the config you sent, I don't see anything about register 
>>>>>>>> expiration, or NAT keepalive, so I would really recommend you 
>>>>>>>> add a ping every 30 seconds to your gateway config.
>>>>>>>>
>>>>>>>> If the issue is there, it's quite easy to see:
>>>>>>>> unregister the gateway
>>>>>>>> register it again
>>>>>>>> make an inbound call right away
>>>>>>>>
>>>>>>>> David Ponzone Direction Technique
>>>>>>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>>>>>>> tel:      01 74 03 18 97
>>>>>>>> gsm:   06 66 98 76 34
>>>>>>>>
>>>>>>>> Service ClientIPeva
>>>>>>>> tel:      0811 46 26 26
>>>>>>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - 
>>>>>>>> www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/>
>>>>>>>>
>>>>>>>> /Ce message et toutes les pi?ces jointes sont confidentiels et 
>>>>>>>> ?tablis ? l'intention exclusive de ses destinataires. Toute 
>>>>>>>> utilisation ou diffusion non autoris?e est interdite. Tout 
>>>>>>>> message ?lectronique est susceptible d'alt?ration. 
>>>>>>>> /*/IPeva/*/ d?cline toute responsabilit? au titre de ce message 
>>>>>>>> s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas 
>>>>>>>> destinataire de ce message, merci de le d?truire imm?diatement 
>>>>>>>> et d'avertir l'exp?diteur./
>>>>>>>> /
>>>>>>>> /
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Le 23/06/2011 ? 18:15, William Moore a ?crit :
>>>>>>>>
>>>>>>>>> It was the 1 and it needs to be set for e.164(+1). Blue.box 
>>>>>>>>> makes it a little more difficult since it wants to manage the 
>>>>>>>>> xmls. I could have set it up as a separate config file however 
>>>>>>>>> that defeats the purpose of the GUI.
>>>>>>>>>
>>>>>>>>> I still have the issue of incoming calls not even showing up 
>>>>>>>>> in sofia. Junction Networks says it is going out, but my 
>>>>>>>>> server is not responding. I'm not sure if I have the settings 
>>>>>>>>> for incoming correct.
>>>>>>>>> William J. Moore
>>>>>>>>> Statira Systems
>>>>>>>>> 611 Caroline St
>>>>>>>>> Fredericksburg, VA
>>>>>>>>> http://www.statirasystems.com
>>>>>>>>>
>>>>>>>>> On 06/23/2011 04:21 AM, David Ponzone wrote:
>>>>>>>>>> William,
>>>>>>>>>>
>>>>>>>>>> If I believe the example configuraiton on Junction's web, 
>>>>>>>>>> they don't expect E164:
>>>>>>>>>>
>>>>>>>>>>     <extension name="junctionnetworks.com  <http://junctionnetworks.com/>>
>>>>>>>>>>      <condition field="destination_number" expression="^(1{0,1}\d{10})$">
>>>>>>>>>>        <action application="set"data="effective_caller_id_number=OUTBOUNDCALLERID"/>
>>>>>>>>>>        <action application="bridge" data="sofia/gateway/junctionnetworks.com/$1"/>
>>>>>>>>>>       </condition>
>>>>>>>>>>     </extension>
>>>>>>>>>> But, elsewhere in their knowledgebase, they say:
>>>>>>>>>>
>>>>>>>>>> *Numbering*
>>>>>>>>>> The standard e.164 numbering plan (ITU <http://www.itu.int/>) 
>>>>>>>>>> is used. North American numbers are required to be prefixed 
>>>>>>>>>> with a '1'. International numbers need to be prefixed with "011".
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> I think they are confused about what is E164....
>>>>>>>>>>
>>>>>>>>>> You should try to prefix with 1 as Michael recommended.
>>>>>>>>>>
>>>>>>>>>> David Ponzone Direction Technique
>>>>>>>>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>>>>>>>>> tel:      01 74 03 18 97
>>>>>>>>>> gsm:   06 66 98 76 34
>>>>>>>>>>
>>>>>>>>>> Service ClientIPeva
>>>>>>>>>> tel:      0811 46 26 26
>>>>>>>>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - 
>>>>>>>>>> www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/>
>>>>>>>>>>
>>>>>>>>>> /Ce message et toutes les pi?ces jointes sont confidentiels 
>>>>>>>>>> et ?tablis ? l'intention exclusive de ses destinataires. 
>>>>>>>>>> Toute utilisation ou diffusion non autoris?e est interdite. 
>>>>>>>>>> Tout message ?lectronique est susceptible d'alt?ration. 
>>>>>>>>>> /*/IPeva/*/ d?cline toute responsabilit? au titre de ce 
>>>>>>>>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
>>>>>>>>>> n'?tes pas destinataire de ce message, merci de le d?truire 
>>>>>>>>>> imm?diatement et d'avertir l'exp?diteur./
>>>>>>>>>> /
>>>>>>>>>> /
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> Le 23/06/2011 ? 08:19, Michael Collins a ?crit :
>>>>>>>>>>
>>>>>>>>>>> Perhaps the gateway wants the leading 1 or expects e.164 
>>>>>>>>>>> format?
>>>>>>>>>>> -MC
>>>>>>>>>>>
>>>>>>>>>>> On Wed, Jun 22, 2011 at 5:37 PM, Brian West 
>>>>>>>>>>> <brian at freeswitch.org <mailto:brian at freeswitch.org>> wrote:
>>>>>>>>>>>
>>>>>>>>>>>     484 is address incomplete means you didn't send enough
>>>>>>>>>>>     digits to the gateway usually.
>>>>>>>>>>>
>>>>>>>>>>>     /b
>>>>>>>>>>>
>>>>>>>>>>>     On Jun 22, 2011, at 7:34 PM, Brad Mina wrote:
>>>>>>>>>>>
>>>>>>>>>>>>     Ah, thought it was recv not send. Good to know that an
>>>>>>>>>>>>     invalid gateway issue
>>>>>>>>>>>>     can cause this.
>>>>>>>>>>>>
>>>>>>>>>>>>     On Wed, Jun 22, 2011 at 5:31 PM, Michael Collins
>>>>>>>>>>>>     <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>>     The 484 I believe is a result of the bad gateway. In
>>>>>>>>>>>>>     any case, the OP needs
>>>>>>>>>>>>>     to figure out what's up w/ that gw or no calls will be
>>>>>>>>>>>>>     going out...
>>>>>>>>>>>>>
>>>>>>>>>>>>>     -MC
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>     _______________________________________________
>>>>>>>>>>>     Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>     http://www.cluecon.com <http://www.cluecon.com/>
>>>>>>>>>>>     877-7-4ACLUE
>>>>>>>>>>>
>>>>>>>>>>>     FreeSWITCH-users mailing list
>>>>>>>>>>>     FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>> http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>>>>>>>>>
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org 
>>>>>>>>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>> http://www.cluecon.com  877-7-4ACLUE
>>>>>>>>>>
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>>>>>>> _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org 
>>>>>>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com  877-7-4ACLUE
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com  877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org 
>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com  877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mehmasarja at gmail.com  Fri Jun 24 19:41:07 2011
From: mehmasarja at gmail.com (Mehma Sarja)
Date: Fri, 24 Jun 2011 08:41:07 -0700
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <4E043741.8090204@puzzled.xs4all.nl>
References: <CA293E4C.18131%krice@freeswitch.org>	<32B9468B-3402-4056-A18D-D4A01626A87A@gmail.com>
	<4E043741.8090204@puzzled.xs4all.nl>
Message-ID: <4E04B013.4010608@gmail.com>

On 6/24/11 12:05 AM, Patrick Lists wrote:
>
> B8ZS line encoding. Iirc a funky version of AMI because the US suffered
> from not-invented-here syndrom so they came up with a tweaked version of
> AMI called Bipolar-8-0-substition. Nuf said :) Brings back memories of
> long nights troubleshooting international circuits with BERT testers and
> hopping on planes to follow the nodes so we could finally pinpoint which
> carrier was lying about nothing being wrong with their stretch.
> erimental DWDM and SDH gear that we got to play with.
>
> "If we build it they will come" - wonder if we will ever see those days
> again.
Very interesting window into history.

Mehma


From bmoore at statirasystems.com  Fri Jun 24 19:41:45 2011
From: bmoore at statirasystems.com (William Moore)
Date: Fri, 24 Jun 2011 11:41:45 -0400
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <CA293ED7.2B6A6%d@d-man.org>
References: <CA293ED7.2B6A6%d@d-man.org>
Message-ID: <4E04B039.6020706@statirasystems.com>

Thanks for responding. I just stayed up all night learning all I could 
about both freeswitch and blue.box. I had a client pushing me so I was a 
little frustrated with it. I usually troll and learn it on my own via 
the code, documentation, bug lists, and so on. I was looking for a more 
immediate response than normal.

I do appreciate the project and all the work put forward by you and your 
team.

The only other software I had as much trouble learning was when I 
switched to Gentoo and found out how little I knew about linux.

You will hear from me again. Hopefully via contributions.

William J. Moore
Statira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com


On 06/23/2011 10:05 PM, Darren Schreiber wrote:
> OK, well again, I'm happy to help, as is my team and as are the 
> open-source folks. Sometimes we do get busy, so please just ping me, 
> but the logs pasted by William indicate he doesn't know how to setup 
> the authentication for his carriers/phones and since there's 82 
> different ways to do that, I'd need to know more info.
>
> Hence the request to email off list, unless he wants to post his auth 
> strategy and IPs publicly :)
>
> -- 
>
>
> From: David Ponzone <david.ponzone at ipeva.fr 
> <mailto:david.ponzone at ipeva.fr>>
> Reply-To: "freeswitch-users at lists.freeswitch.org 
> <mailto:freeswitch-users at lists.freeswitch.org>" 
> <freeswitch-users at lists.freeswitch.org 
> <mailto:freeswitch-users at lists.freeswitch.org>>
> Date: Thu, 23 Jun 2011 18:56:04 -0700
> To: "freeswitch-users at lists.freeswitch.org 
> <mailto:freeswitch-users at lists.freeswitch.org>" 
> <freeswitch-users at lists.freeswitch.org 
> <mailto:freeswitch-users at lists.freeswitch.org>>
> Subject: Re: [Freeswitch-users] Trunk Issue
>
> No Darren, sorry, I confused FreePBX and FusionPBX.
>
> My bad.
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service ClientIPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
> diffusion non autoris?e est interdite. Tout message ?lectronique est 
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au 
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
> n'?tes pas destinataire de ce message, merci de le d?truire 
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 24/06/2011 ? 03:14, Darren Schreiber a ?crit :
>
>> David,
>> FreePBX v3 is now called blue.box. You're not making any sense? Is 
>> this an attempt at creating FUD?
>>
>> - Darren
>>
>> -- 
>>
>>
>> From: David Ponzone <david.ponzone at ipeva.fr 
>> <mailto:david.ponzone at ipeva.fr>>
>> Reply-To: "freeswitch-users at lists.freeswitch.org 
>> <mailto:freeswitch-users at lists.freeswitch.org>" 
>> <freeswitch-users at lists.freeswitch.org 
>> <mailto:freeswitch-users at lists.freeswitch.org>>
>> Date: Thu, 23 Jun 2011 12:59:16 -0700
>> To: "freeswitch-users at lists.freeswitch.org 
>> <mailto:freeswitch-users at lists.freeswitch.org>" 
>> <freeswitch-users at lists.freeswitch.org 
>> <mailto:freeswitch-users at lists.freeswitch.org>>
>> Subject: Re: [Freeswitch-users] Trunk Issue
>>
>> I had the same feeling about blue.box.
>> I think there are more interesting GUIs out there, like FreePBX V3.
>> The weird thing is that none of them seems very active.
>>
>> David Ponzone Direction Technique
>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service ClientIPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
>> <BLOCKED::http://www.ipeva-studio.com/>
>>
>> /Ce message et toutes les pi?ces jointes sont confidentiels et 
>> ?tablis ? l'intention exclusive de ses destinataires. Toute 
>> utilisation ou diffusion non autoris?e est interdite. Tout message 
>> ?lectronique est susceptible d'alt?ration. /*/IPeva/*/ d?cline toute 
>> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou 
>> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le 
>> d?truire imm?diatement et d'avertir l'exp?diteur./
>> /
>> /
>>
>>
>>
>> Le 23/06/2011 ? 19:55, William Moore a ?crit :
>>
>>> I give up on blue.box. They seem preoccupied by there other 
>>> projects. Documentation does not explain enough. It is like 
>>> describing a human by their parts only and not how it interacts. 
>>> Peace out.
>>> William J. Moore
>>> Statira Systems
>>> 611 Caroline St
>>> Fredericksburg, VA
>>> http://www.statirasystems.com
>>>
>>> On 06/23/2011 01:34 PM, David Ponzone wrote:
>>>> In sipinterface_2, make the following modification:
>>>>
>>>> <param name="auth-calls" value="false"/>
>>>>
>>>>
>>>> David Ponzone Direction Technique
>>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>>> tel:      01 74 03 18 97
>>>> gsm:   06 66 98 76 34
>>>>
>>>> Service ClientIPeva
>>>> tel:      0811 46 26 26
>>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - 
>>>> www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/>
>>>>
>>>> /Ce message et toutes les pi?ces jointes sont confidentiels et 
>>>> ?tablis ? l'intention exclusive de ses destinataires. Toute 
>>>> utilisation ou diffusion non autoris?e est interdite. Tout message 
>>>> ?lectronique est susceptible d'alt?ration. /*/IPeva/*/ d?cline 
>>>> toute responsabilit? au titre de ce message s'il a ?t? alt?r?, 
>>>> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, 
>>>> merci de le d?truire imm?diatement et d'avertir l'exp?diteur./
>>>> /
>>>> /
>>>>
>>>>
>>>>
>>>> Le 23/06/2011 ? 19:31, David Ponzone a ?crit :
>>>>
>>>>> The issue is that you are expecting them to authenticate, but they 
>>>>> won't.
>>>>>
>>>>> David Ponzone Direction Technique
>>>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>>>> tel:      01 74 03 18 97
>>>>> gsm:   06 66 98 76 34
>>>>>
>>>>> Service ClientIPeva
>>>>> tel:      0811 46 26 26
>>>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - 
>>>>> www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/>
>>>>>
>>>>> /Ce message et toutes les pi?ces jointes sont confidentiels et 
>>>>> ?tablis ? l'intention exclusive de ses destinataires. Toute 
>>>>> utilisation ou diffusion non autoris?e est interdite. Tout message 
>>>>> ?lectronique est susceptible d'alt?ration. /*/IPeva/*/ d?cline 
>>>>> toute responsabilit? au titre de ce message s'il a ?t? alt?r?, 
>>>>> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce 
>>>>> message, merci de le d?truire imm?diatement et d'avertir 
>>>>> l'exp?diteur./
>>>>> /
>>>>> /
>>>>>
>>>>>
>>>>>
>>>>> Le 23/06/2011 ? 19:15, William Moore a ?crit :
>>>>>
>>>>>> Ok, I reregistered and called right away. I get the following:
>>>>>>    0.000000 66.227.100.20 ->  192.168.1.254 SIP/SDP Request: INVITEsip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with session description
>>>>>>    0.000766 192.168.1.254 ->  66.227.100.20 SIP Status: 100 Trying
>>>>>>    0.002321 192.168.1.254 ->  66.227.100.20 SIP Status: 407 Proxy Authentication Required
>>>>>>    0.021120 66.227.100.20 ->  192.168.1.254 SIP Request: ACKsip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>>>>>    3.272168 192.168.1.254 ->  66.227.100.20 SIP Request: REGISTERsip:jnctn.net;transport=udp
>>>>>>    3.292076 66.227.100.20 ->  192.168.1.254 SIP Status: 200 OK    (1 bindings)
>>>>>>
>>>>>> The call just states it is a none working number. That means something in my dial plan is incorrect. I am double checking it now.
>>>>>> William J. MooreStatira Systems
>>>>>> 611 Caroline St
>>>>>> Fredericksburg, VA
>>>>>> http://www.statirasystems.com
>>>>>>
>>>>>> On 06/23/2011 12:58 PM, William Moore wrote:
>>>>>>> When I put it in and it reloads, blue.box removes it. Can I 
>>>>>>> reference the gateway in another sip interface file or does it 
>>>>>>> have to be in that one?
>>>>>>> William J. Moore
>>>>>>> Statira Systems
>>>>>>> 611 Caroline St
>>>>>>> Fredericksburg, VA
>>>>>>> http://www.statirasystems.com
>>>>>>>
>>>>>>> On 06/23/2011 12:52 PM, William Moore wrote:
>>>>>>>> correction: missed a line.
>>>>>>>>
>>>>>>>>   0.000000 66.227.100.20 -> 192.168.1.254 SIP/SDP Request: 
>>>>>>>> INVITE 
>>>>>>>> sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, 
>>>>>>>> with session description
>>>>>>>>   0.000814 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>>>>>>>>   0.002440 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy 
>>>>>>>> Authentication Required
>>>>>>>>   0.023173 66.227.100.20 -> 192.168.1.254 SIP Request: ACK 
>>>>>>>> sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>>>>>>>
>>>>>>>> William J. Moore
>>>>>>>> Statira Systems
>>>>>>>> 611 Caroline St
>>>>>>>> Fredericksburg, VA
>>>>>>>> http://www.statirasystems.com
>>>>>>>>
>>>>>>>> On 06/23/2011 12:50 PM, William Moore wrote:
>>>>>>>>> I get this with wireshark
>>>>>>>>>
>>>>>>>>>  40.345679 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>>>>>>>>>  40.347381 192.168.1.254 -> 66.227.100.20 SIP Status: 407 
>>>>>>>>> Proxy Authentication Required
>>>>>>>>>  40.367906 66.227.100.20 -> 192.168.1.254 SIP Request: ACK 
>>>>>>>>> sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>>>>>>>>
>>>>>>>>> I am looking for the expiration now..
>>>>>>>>> William J. Moore
>>>>>>>>> Statira Systems
>>>>>>>>> 611 Caroline St
>>>>>>>>> Fredericksburg, VA
>>>>>>>>> http://www.statirasystems.com
>>>>>>>>>
>>>>>>>>> On 06/23/2011 12:25 PM, David Ponzone wrote:
>>>>>>>>>> Take a sip trace on your box with tcpdump or preferably, tshark:
>>>>>>>>>> tshark port 5080
>>>>>>>>>>
>>>>>>>>>> You will then see if you receive it.
>>>>>>>>>>
>>>>>>>>>> In the config you sent, I don't see anything about register 
>>>>>>>>>> expiration, or NAT keepalive, so I would really recommend you 
>>>>>>>>>> add a ping every 30 seconds to your gateway config.
>>>>>>>>>>
>>>>>>>>>> If the issue is there, it's quite easy to see:
>>>>>>>>>> unregister the gateway
>>>>>>>>>> register it again
>>>>>>>>>> make an inbound call right away
>>>>>>>>>>
>>>>>>>>>> David Ponzone Direction Technique
>>>>>>>>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>>>>>>>>> tel:      01 74 03 18 97
>>>>>>>>>> gsm:   06 66 98 76 34
>>>>>>>>>>
>>>>>>>>>> Service ClientIPeva
>>>>>>>>>> tel:      0811 46 26 26
>>>>>>>>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - 
>>>>>>>>>> www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/>
>>>>>>>>>>
>>>>>>>>>> /Ce message et toutes les pi?ces jointes sont confidentiels 
>>>>>>>>>> et ?tablis ? l'intention exclusive de ses destinataires. 
>>>>>>>>>> Toute utilisation ou diffusion non autoris?e est interdite. 
>>>>>>>>>> Tout message ?lectronique est susceptible d'alt?ration. 
>>>>>>>>>> /*/IPeva/*/ d?cline toute responsabilit? au titre de ce 
>>>>>>>>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
>>>>>>>>>> n'?tes pas destinataire de ce message, merci de le d?truire 
>>>>>>>>>> imm?diatement et d'avertir l'exp?diteur./
>>>>>>>>>> /
>>>>>>>>>> /
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> Le 23/06/2011 ? 18:15, William Moore a ?crit :
>>>>>>>>>>
>>>>>>>>>>> It was the 1 and it needs to be set for e.164(+1). Blue.box 
>>>>>>>>>>> makes it a little more difficult since it wants to manage 
>>>>>>>>>>> the xmls. I could have set it up as a separate config file 
>>>>>>>>>>> however that defeats the purpose of the GUI.
>>>>>>>>>>>
>>>>>>>>>>> I still have the issue of incoming calls not even showing up 
>>>>>>>>>>> in sofia. Junction Networks says it is going out, but my 
>>>>>>>>>>> server is not responding. I'm not sure if I have the 
>>>>>>>>>>> settings for incoming correct.
>>>>>>>>>>> William J. Moore
>>>>>>>>>>> Statira Systems
>>>>>>>>>>> 611 Caroline St
>>>>>>>>>>> Fredericksburg, VA
>>>>>>>>>>> http://www.statirasystems.com
>>>>>>>>>>>
>>>>>>>>>>> On 06/23/2011 04:21 AM, David Ponzone wrote:
>>>>>>>>>>>> William,
>>>>>>>>>>>>
>>>>>>>>>>>> If I believe the example configuraiton on Junction's web, 
>>>>>>>>>>>> they don't expect E164:
>>>>>>>>>>>>
>>>>>>>>>>>>     <extension name="junctionnetworks.com  <http://junctionnetworks.com/>>
>>>>>>>>>>>>      <condition field="destination_number" expression="^(1{0,1}\d{10})$">
>>>>>>>>>>>>        <action application="set"data="effective_caller_id_number=OUTBOUNDCALLERID"/>
>>>>>>>>>>>>        <action application="bridge" data="sofia/gateway/junctionnetworks.com/$1"/>
>>>>>>>>>>>>       </condition>
>>>>>>>>>>>>     </extension>
>>>>>>>>>>>> But, elsewhere in their knowledgebase, they say:
>>>>>>>>>>>>
>>>>>>>>>>>> *Numbering*
>>>>>>>>>>>> The standard e.164 numbering plan (ITU 
>>>>>>>>>>>> <http://www.itu.int/>) is used. North American numbers are 
>>>>>>>>>>>> required to be prefixed with a '1'. International numbers 
>>>>>>>>>>>> need to be prefixed with "011".
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> I think they are confused about what is E164....
>>>>>>>>>>>>
>>>>>>>>>>>> You should try to prefix with 1 as Michael recommended.
>>>>>>>>>>>>
>>>>>>>>>>>> David Ponzone Direction Technique
>>>>>>>>>>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>>>>>>>>>>> tel:      01 74 03 18 97
>>>>>>>>>>>> gsm:   06 66 98 76 34
>>>>>>>>>>>>
>>>>>>>>>>>> Service ClientIPeva
>>>>>>>>>>>> tel:      0811 46 26 26
>>>>>>>>>>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - 
>>>>>>>>>>>> www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/>
>>>>>>>>>>>>
>>>>>>>>>>>> /Ce message et toutes les pi?ces jointes sont confidentiels 
>>>>>>>>>>>> et ?tablis ? l'intention exclusive de ses destinataires. 
>>>>>>>>>>>> Toute utilisation ou diffusion non autoris?e est interdite. 
>>>>>>>>>>>> Tout message ?lectronique est susceptible d'alt?ration. 
>>>>>>>>>>>> /*/IPeva/*/ d?cline toute responsabilit? au titre de ce 
>>>>>>>>>>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
>>>>>>>>>>>> n'?tes pas destinataire de ce message, merci de le d?truire 
>>>>>>>>>>>> imm?diatement et d'avertir l'exp?diteur./
>>>>>>>>>>>> /
>>>>>>>>>>>> /
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> Le 23/06/2011 ? 08:19, Michael Collins a ?crit :
>>>>>>>>>>>>
>>>>>>>>>>>>> Perhaps the gateway wants the leading 1 or expects e.164 
>>>>>>>>>>>>> format?
>>>>>>>>>>>>> -MC
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Wed, Jun 22, 2011 at 5:37 PM, Brian West 
>>>>>>>>>>>>> <brian at freeswitch.org <mailto:brian at freeswitch.org>> wrote:
>>>>>>>>>>>>>> 484 is address incomplete means you didn't send enough 
>>>>>>>>>>>>>> digits to the gateway usually.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> /b
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> On Jun 22, 2011, at 7:34 PM, Brad Mina wrote:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Ah, thought it was recv not send. Good to know that an 
>>>>>>>>>>>>>>> invalid gateway issue
>>>>>>>>>>>>>>> can cause this.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> On Wed, Jun 22, 2011 at 5:31 PM, Michael Collins 
>>>>>>>>>>>>>>> <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> The 484 I believe is a result of the bad gateway. In 
>>>>>>>>>>>>>>>> any case, the OP needs
>>>>>>>>>>>>>>>> to figure out what's up w/ that gw or no calls will be 
>>>>>>>>>>>>>>>> going out...
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> -MC
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>>>> http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org 
>>>>>>>>>>>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>>> http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>>>>>>>>>>>
>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org 
>>>>>>>>>>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>> http://www.cluecon.com  877-7-4ACLUE
>>>>>>>>>>>>
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>> http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>>>>>>>>>
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org 
>>>>>>>>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>> http://www.cluecon.com  877-7-4ACLUE
>>>>>>>>>>
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com  877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org 
>>>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com  877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org 
>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org 
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From bmoore at statirasystems.com  Fri Jun 24 19:52:39 2011
From: bmoore at statirasystems.com (William Moore)
Date: Fri, 24 Jun 2011 11:52:39 -0400
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <4E04B039.6020706@statirasystems.com>
References: <CA293ED7.2B6A6%d@d-man.org> <4E04B039.6020706@statirasystems.com>
Message-ID: <4E04B2C7.7060906@statirasystems.com>

And yes I got it all working via wireshark and sofia siptrace.

Primary issue for incoming was a 1 at the beginning of the number and 
the loop you mentioned. As you probably saw there were a few other 
errors in the settings.

William J. Moore
Statira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com


On 06/24/2011 11:41 AM, William Moore wrote:
> Thanks for responding. I just stayed up all night learning all I could 
> about both freeswitch and blue.box. I had a client pushing me so I was 
> a little frustrated with it. I usually troll and learn it on my own 
> via the code, documentation, bug lists, and so on. I was looking for a 
> more immediate response than normal.
>
> I do appreciate the project and all the work put forward by you and 
> your team.
>
> The only other software I had as much trouble learning was when I 
> switched to Gentoo and found out how little I knew about linux.
>
> You will hear from me again. Hopefully via contributions.
> William J. Moore
> Statira Systems
> 611 Caroline St
> Fredericksburg, VA
> http://www.statirasystems.com
>
> On 06/23/2011 10:05 PM, Darren Schreiber wrote:
>> OK, well again, I'm happy to help, as is my team and as are the 
>> open-source folks. Sometimes we do get busy, so please just ping me, 
>> but the logs pasted by William indicate he doesn't know how to setup 
>> the authentication for his carriers/phones and since there's 82 
>> different ways to do that, I'd need to know more info.
>>
>> Hence the request to email off list, unless he wants to post his auth 
>> strategy and IPs publicly :)
>>
>> -- 
>>
>>
>> From: David Ponzone <david.ponzone at ipeva.fr 
>> <mailto:david.ponzone at ipeva.fr>>
>> Reply-To: "freeswitch-users at lists.freeswitch.org 
>> <mailto:freeswitch-users at lists.freeswitch.org>" 
>> <freeswitch-users at lists.freeswitch.org 
>> <mailto:freeswitch-users at lists.freeswitch.org>>
>> Date: Thu, 23 Jun 2011 18:56:04 -0700
>> To: "freeswitch-users at lists.freeswitch.org 
>> <mailto:freeswitch-users at lists.freeswitch.org>" 
>> <freeswitch-users at lists.freeswitch.org 
>> <mailto:freeswitch-users at lists.freeswitch.org>>
>> Subject: Re: [Freeswitch-users] Trunk Issue
>>
>> No Darren, sorry, I confused FreePBX and FusionPBX.
>>
>> My bad.
>>
>> David Ponzone Direction Technique
>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service ClientIPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
>> <BLOCKED::http://www.ipeva-studio.com/>
>>
>> /Ce message et toutes les pi?ces jointes sont confidentiels et 
>> ?tablis ? l'intention exclusive de ses destinataires. Toute 
>> utilisation ou diffusion non autoris?e est interdite. Tout message 
>> ?lectronique est susceptible d'alt?ration. /*/IPeva/*/ d?cline toute 
>> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou 
>> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le 
>> d?truire imm?diatement et d'avertir l'exp?diteur./
>> /
>> /
>>
>>
>>
>> Le 24/06/2011 ? 03:14, Darren Schreiber a ?crit :
>>
>>> David,
>>> FreePBX v3 is now called blue.box. You're not making any sense? Is 
>>> this an attempt at creating FUD?
>>>
>>> - Darren
>>>
>>> -- 
>>>
>>>
>>> From: David Ponzone <david.ponzone at ipeva.fr 
>>> <mailto:david.ponzone at ipeva.fr>>
>>> Reply-To: "freeswitch-users at lists.freeswitch.org 
>>> <mailto:freeswitch-users at lists.freeswitch.org>" 
>>> <freeswitch-users at lists.freeswitch.org 
>>> <mailto:freeswitch-users at lists.freeswitch.org>>
>>> Date: Thu, 23 Jun 2011 12:59:16 -0700
>>> To: "freeswitch-users at lists.freeswitch.org 
>>> <mailto:freeswitch-users at lists.freeswitch.org>" 
>>> <freeswitch-users at lists.freeswitch.org 
>>> <mailto:freeswitch-users at lists.freeswitch.org>>
>>> Subject: Re: [Freeswitch-users] Trunk Issue
>>>
>>> I had the same feeling about blue.box.
>>> I think there are more interesting GUIs out there, like FreePBX V3.
>>> The weird thing is that none of them seems very active.
>>>
>>> David Ponzone Direction Technique
>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service ClientIPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
>>> <BLOCKED::http://www.ipeva-studio.com/>
>>>
>>> /Ce message et toutes les pi?ces jointes sont confidentiels et 
>>> ?tablis ? l'intention exclusive de ses destinataires. Toute 
>>> utilisation ou diffusion non autoris?e est interdite. Tout message 
>>> ?lectronique est susceptible d'alt?ration. /*/IPeva/*/ d?cline toute 
>>> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou 
>>> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le 
>>> d?truire imm?diatement et d'avertir l'exp?diteur./
>>> /
>>> /
>>>
>>>
>>>
>>> Le 23/06/2011 ? 19:55, William Moore a ?crit :
>>>
>>>> I give up on blue.box. They seem preoccupied by there other 
>>>> projects. Documentation does not explain enough. It is like 
>>>> describing a human by their parts only and not how it interacts. 
>>>> Peace out.
>>>> William J. Moore
>>>> Statira Systems
>>>> 611 Caroline St
>>>> Fredericksburg, VA
>>>> http://www.statirasystems.com
>>>>
>>>> On 06/23/2011 01:34 PM, David Ponzone wrote:
>>>>> In sipinterface_2, make the following modification:
>>>>>
>>>>> <param name="auth-calls" value="false"/>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> David Ponzone Direction Technique
>>>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>>>> tel:      01 74 03 18 97
>>>>> gsm:   06 66 98 76 34
>>>>>
>>>>> Service ClientIPeva
>>>>> tel:      0811 46 26 26
>>>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - 
>>>>> www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/>
>>>>>
>>>>> /Ce message et toutes les pi?ces jointes sont confidentiels et 
>>>>> ?tablis ? l'intention exclusive de ses destinataires. Toute 
>>>>> utilisation ou diffusion non autoris?e est interdite. Tout message 
>>>>> ?lectronique est susceptible d'alt?ration. /*/IPeva/*/ d?cline 
>>>>> toute responsabilit? au titre de ce message s'il a ?t? alt?r?, 
>>>>> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce 
>>>>> message, merci de le d?truire imm?diatement et d'avertir 
>>>>> l'exp?diteur./
>>>>> /
>>>>> /
>>>>>
>>>>>
>>>>>
>>>>> Le 23/06/2011 ? 19:31, David Ponzone a ?crit :
>>>>>
>>>>>> The issue is that you are expecting them to authenticate, but 
>>>>>> they won't.
>>>>>>
>>>>>> David Ponzone Direction Technique
>>>>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>>>>> tel:      01 74 03 18 97
>>>>>> gsm:   06 66 98 76 34
>>>>>>
>>>>>> Service ClientIPeva
>>>>>> tel:      0811 46 26 26
>>>>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - 
>>>>>> www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/>
>>>>>>
>>>>>> /Ce message et toutes les pi?ces jointes sont confidentiels et 
>>>>>> ?tablis ? l'intention exclusive de ses destinataires. Toute 
>>>>>> utilisation ou diffusion non autoris?e est interdite. Tout 
>>>>>> message ?lectronique est susceptible d'alt?ration. 
>>>>>> /*/IPeva/*/ d?cline toute responsabilit? au titre de ce message 
>>>>>> s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas 
>>>>>> destinataire de ce message, merci de le d?truire imm?diatement et 
>>>>>> d'avertir l'exp?diteur./
>>>>>> /
>>>>>> /
>>>>>>
>>>>>>
>>>>>>
>>>>>> Le 23/06/2011 ? 19:15, William Moore a ?crit :
>>>>>>
>>>>>>> Ok, I reregistered and called right away. I get the following:
>>>>>>>    0.000000 66.227.100.20 ->  192.168.1.254 SIP/SDP Request: INVITEsip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with session description
>>>>>>>    0.000766 192.168.1.254 ->  66.227.100.20 SIP Status: 100 Trying
>>>>>>>    0.002321 192.168.1.254 ->  66.227.100.20 SIP Status: 407 Proxy Authentication Required
>>>>>>>    0.021120 66.227.100.20 ->  192.168.1.254 SIP Request: ACKsip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>>>>>>    3.272168 192.168.1.254 ->  66.227.100.20 SIP Request: REGISTERsip:jnctn.net;transport=udp
>>>>>>>    3.292076 66.227.100.20 ->  192.168.1.254 SIP Status: 200 OK    (1 bindings)
>>>>>>>
>>>>>>> The call just states it is a none working number. That means something in my dial plan is incorrect. I am double checking it now.
>>>>>>> William J. MooreStatira Systems
>>>>>>> 611 Caroline St
>>>>>>> Fredericksburg, VA
>>>>>>> http://www.statirasystems.com
>>>>>>>
>>>>>>> On 06/23/2011 12:58 PM, William Moore wrote:
>>>>>>>> When I put it in and it reloads, blue.box removes it. Can I 
>>>>>>>> reference the gateway in another sip interface file or does it 
>>>>>>>> have to be in that one?
>>>>>>>> William J. Moore
>>>>>>>> Statira Systems
>>>>>>>> 611 Caroline St
>>>>>>>> Fredericksburg, VA
>>>>>>>> http://www.statirasystems.com
>>>>>>>>
>>>>>>>> On 06/23/2011 12:52 PM, William Moore wrote:
>>>>>>>>> correction: missed a line.
>>>>>>>>>
>>>>>>>>>   0.000000 66.227.100.20 -> 192.168.1.254 SIP/SDP Request: 
>>>>>>>>> INVITE 
>>>>>>>>> sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, 
>>>>>>>>> with session description
>>>>>>>>>   0.000814 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>>>>>>>>>   0.002440 192.168.1.254 -> 66.227.100.20 SIP Status: 407 
>>>>>>>>> Proxy Authentication Required
>>>>>>>>>   0.023173 66.227.100.20 -> 192.168.1.254 SIP Request: ACK 
>>>>>>>>> sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>>>>>>>>
>>>>>>>>> William J. Moore
>>>>>>>>> Statira Systems
>>>>>>>>> 611 Caroline St
>>>>>>>>> Fredericksburg, VA
>>>>>>>>> http://www.statirasystems.com
>>>>>>>>>
>>>>>>>>> On 06/23/2011 12:50 PM, William Moore wrote:
>>>>>>>>>> I get this with wireshark
>>>>>>>>>>
>>>>>>>>>>  40.345679 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
>>>>>>>>>>  40.347381 192.168.1.254 -> 66.227.100.20 SIP Status: 407 
>>>>>>>>>> Proxy Authentication Required
>>>>>>>>>>  40.367906 66.227.100.20 -> 192.168.1.254 SIP Request: ACK 
>>>>>>>>>> sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2
>>>>>>>>>>
>>>>>>>>>> I am looking for the expiration now..
>>>>>>>>>> William J. Moore
>>>>>>>>>> Statira Systems
>>>>>>>>>> 611 Caroline St
>>>>>>>>>> Fredericksburg, VA
>>>>>>>>>> http://www.statirasystems.com
>>>>>>>>>>
>>>>>>>>>> On 06/23/2011 12:25 PM, David Ponzone wrote:
>>>>>>>>>>> Take a sip trace on your box with tcpdump or preferably, 
>>>>>>>>>>> tshark:
>>>>>>>>>>> tshark port 5080
>>>>>>>>>>>
>>>>>>>>>>> You will then see if you receive it.
>>>>>>>>>>>
>>>>>>>>>>> In the config you sent, I don't see anything about register 
>>>>>>>>>>> expiration, or NAT keepalive, so I would really recommend 
>>>>>>>>>>> you add a ping every 30 seconds to your gateway config.
>>>>>>>>>>>
>>>>>>>>>>> If the issue is there, it's quite easy to see:
>>>>>>>>>>> unregister the gateway
>>>>>>>>>>> register it again
>>>>>>>>>>> make an inbound call right away
>>>>>>>>>>>
>>>>>>>>>>> David Ponzone Direction Technique
>>>>>>>>>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>>>>>>>>>> tel:      01 74 03 18 97
>>>>>>>>>>> gsm:   06 66 98 76 34
>>>>>>>>>>>
>>>>>>>>>>> Service ClientIPeva
>>>>>>>>>>> tel:      0811 46 26 26
>>>>>>>>>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - 
>>>>>>>>>>> www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/>
>>>>>>>>>>>
>>>>>>>>>>> /Ce message et toutes les pi?ces jointes sont confidentiels 
>>>>>>>>>>> et ?tablis ? l'intention exclusive de ses destinataires. 
>>>>>>>>>>> Toute utilisation ou diffusion non autoris?e est interdite. 
>>>>>>>>>>> Tout message ?lectronique est susceptible d'alt?ration. 
>>>>>>>>>>> /*/IPeva/*/ d?cline toute responsabilit? au titre de ce 
>>>>>>>>>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
>>>>>>>>>>> n'?tes pas destinataire de ce message, merci de le d?truire 
>>>>>>>>>>> imm?diatement et d'avertir l'exp?diteur./
>>>>>>>>>>> /
>>>>>>>>>>> /
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> Le 23/06/2011 ? 18:15, William Moore a ?crit :
>>>>>>>>>>>
>>>>>>>>>>>> It was the 1 and it needs to be set for e.164(+1). Blue.box 
>>>>>>>>>>>> makes it a little more difficult since it wants to manage 
>>>>>>>>>>>> the xmls. I could have set it up as a separate config file 
>>>>>>>>>>>> however that defeats the purpose of the GUI.
>>>>>>>>>>>>
>>>>>>>>>>>> I still have the issue of incoming calls not even showing 
>>>>>>>>>>>> up in sofia. Junction Networks says it is going out, but my 
>>>>>>>>>>>> server is not responding. I'm not sure if I have the 
>>>>>>>>>>>> settings for incoming correct.
>>>>>>>>>>>> William J. Moore
>>>>>>>>>>>> Statira Systems
>>>>>>>>>>>> 611 Caroline St
>>>>>>>>>>>> Fredericksburg, VA
>>>>>>>>>>>> http://www.statirasystems.com
>>>>>>>>>>>>
>>>>>>>>>>>> On 06/23/2011 04:21 AM, David Ponzone wrote:
>>>>>>>>>>>>> William,
>>>>>>>>>>>>>
>>>>>>>>>>>>> If I believe the example configuraiton on Junction's web, 
>>>>>>>>>>>>> they don't expect E164:
>>>>>>>>>>>>>
>>>>>>>>>>>>>     <extension name="junctionnetworks.com  <http://junctionnetworks.com/>>
>>>>>>>>>>>>>      <condition field="destination_number" expression="^(1{0,1}\d{10})$">
>>>>>>>>>>>>>        <action application="set"data="effective_caller_id_number=OUTBOUNDCALLERID"/>
>>>>>>>>>>>>>        <action application="bridge" data="sofia/gateway/junctionnetworks.com/$1"/>
>>>>>>>>>>>>>       </condition>
>>>>>>>>>>>>>     </extension>
>>>>>>>>>>>>> But, elsewhere in their knowledgebase, they say:
>>>>>>>>>>>>>
>>>>>>>>>>>>> *Numbering*
>>>>>>>>>>>>> The standard e.164 numbering plan (ITU 
>>>>>>>>>>>>> <http://www.itu.int/>) is used. North American numbers are 
>>>>>>>>>>>>> required to be prefixed with a '1'. International numbers 
>>>>>>>>>>>>> need to be prefixed with "011".
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> I think they are confused about what is E164....
>>>>>>>>>>>>>
>>>>>>>>>>>>> You should try to prefix with 1 as Michael recommended.
>>>>>>>>>>>>>
>>>>>>>>>>>>> David Ponzone Direction Technique
>>>>>>>>>>>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>>>>>>>>>>>> tel:      01 74 03 18 97
>>>>>>>>>>>>> gsm:   06 66 98 76 34
>>>>>>>>>>>>>
>>>>>>>>>>>>> Service ClientIPeva
>>>>>>>>>>>>> tel:      0811 46 26 26
>>>>>>>>>>>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - 
>>>>>>>>>>>>> www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/>
>>>>>>>>>>>>>
>>>>>>>>>>>>> /Ce message et toutes les pi?ces jointes sont 
>>>>>>>>>>>>> confidentiels et ?tablis ? l'intention exclusive de ses 
>>>>>>>>>>>>> destinataires. Toute utilisation ou diffusion non 
>>>>>>>>>>>>> autoris?e est interdite. Tout message ?lectronique est 
>>>>>>>>>>>>> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute 
>>>>>>>>>>>>> responsabilit? au titre de ce message s'il a ?t? alt?r?, 
>>>>>>>>>>>>> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce 
>>>>>>>>>>>>> message, merci de le d?truire imm?diatement et d'avertir 
>>>>>>>>>>>>> l'exp?diteur./
>>>>>>>>>>>>> /
>>>>>>>>>>>>> /
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> Le 23/06/2011 ? 08:19, Michael Collins a ?crit :
>>>>>>>>>>>>>
>>>>>>>>>>>>>> Perhaps the gateway wants the leading 1 or expects e.164 
>>>>>>>>>>>>>> format?
>>>>>>>>>>>>>> -MC
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> On Wed, Jun 22, 2011 at 5:37 PM, Brian West 
>>>>>>>>>>>>>> <brian at freeswitch.org <mailto:brian at freeswitch.org>> wrote:
>>>>>>>>>>>>>>> 484 is address incomplete means you didn't send enough 
>>>>>>>>>>>>>>> digits to the gateway usually.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> /b
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> On Jun 22, 2011, at 7:34 PM, Brad Mina wrote:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Ah, thought it was recv not send. Good to know that an 
>>>>>>>>>>>>>>>> invalid gateway issue
>>>>>>>>>>>>>>>> can cause this.
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> On Wed, Jun 22, 2011 at 5:31 PM, Michael Collins 
>>>>>>>>>>>>>>>> <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> The 484 I believe is a result of the bad gateway. In 
>>>>>>>>>>>>>>>>> any case, the OP needs
>>>>>>>>>>>>>>>>> to figure out what's up w/ that gw or no calls will be 
>>>>>>>>>>>>>>>>> going out...
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> -MC
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>>>>> http://www.cluecon.com <http://www.cluecon.com/> 
>>>>>>>>>>>>>>> 877-7-4ACLUE
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org 
>>>>>>>>>>>>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>>>> http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org 
>>>>>>>>>>>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>>> http://www.cluecon.com  877-7-4ACLUE
>>>>>>>>>>>>>
>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>> http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>>>>>>>>>>
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org 
>>>>>>>>>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>> http://www.cluecon.com  877-7-4ACLUE
>>>>>>>>>>>
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com  877-7-4ACLUE
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org 
>>>>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com  877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
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>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org 
>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
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>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com  877-7-4ACLUE
>>
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>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
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From wes-fs at 499x.com  Fri Jun 24 19:57:37 2011
From: wes-fs at 499x.com (Wes)
Date: Fri, 24 Jun 2011 10:57:37 -0500
Subject: [Freeswitch-users] how to use session:playAndGetDigits with the
 "say" command in lua script?
In-Reply-To: <BANLkTi=RHa1W7qxXbQ4MTxhN22Vds97zyA@mail.gmail.com>
References: <4E02484F.90907@499x.com>	<BANLkTikXmE0V6yxfM-HMUQ0FKOrY=K=GOw@mail.gmail.com>	<4E025B13.90701@499x.com>
	<4E02681A.2070703@499x.com>
	<BANLkTi=RHa1W7qxXbQ4MTxhN22Vds97zyA@mail.gmail.com>
Message-ID: <4E04B3F1.7020201@499x.com>

Thanks!... I took a look, and got one working like this:

<include> <!--This line will be ignored it's here to validate the xml 
and is optional -->
<macro name="press_x_to_continue">
<input pattern="^(\d+)$">
<match>
<action function="play-file" data="voicemail/vm-press.wav"/>
<action function="say" data="$1" method="pronounced" type="items"/>
<action function="play-file" data="voicemail/vm-continue.wav"/>
<action function="speak-text" data="yo, you must press $1 to continue, 
do it now"/>
</match>
</input>
</macro>
</include>

but when it got to the speak-text function, it gave this error:

2011-06-24 10:54:40.684627 [ERR] switch_core_speech.c:61 Invalid speech 
module [cepstral]!
2011-06-24 10:54:40.684627 [ERR] switch_ivr_play_say.c:2377 Invalid TTS 
module!

I have the flite engine active, so how do I configure speak-text to try 
using the flite engine instead of cepstral?


On 6/22/2011 6:42 PM, Michael Collins wrote:
> Have you researched the subject of "phrase macros"? That may be your 
> golden ticket. If you have the FS book then look at the latter half of 
> ch 6. Otherwise check out these resources:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_phrase
> conf/lang/en/vm/sounds.xml
> conf/lang/en/vm/tts.xml
>
> The voicemail system uses phrase macros a ton, and effectively, I 
> might add. In short, phrase macros let you piece together sound 
> prompts, calls to the say app, calls to the speak app, calls to the 
> sleep app, etc. I think you would be most interested in the 
> "speak-text" action.
>
> -MC
>
>
> On Wed, Jun 22, 2011 at 3:09 PM, Wes <wes-fs at 499x.com 
> <mailto:wes-fs at 499x.com>> wrote:
>
>     or maybe we just need a new version of this method called
>     "speakAndGetDigits" that specifically takes a couple of string
>     parameters for the words to speak, instead of taking the location
>     of the wav files...
>
>
>     On 6/22/2011 4:13 PM, Wes wrote:
>>     Sorry, I was thrown off by the dialplan example... I'm trying to
>>     use speak... I changed my example to try the speak command and I
>>     still have the same issue with the quotes.
>>
>>     Is it possible to send in a "speak" phrase as the phrase to use
>>     in the PAGD command?  Or do I have to record a wav file and pass
>>     the name of the wave file in for that parameter?
>>
>>     On 6/22/2011 3:51 PM, Michael Collins wrote:
>>>     Can you confirm: are you trying to use TTS or the say engine?
>>>     These are two completely different subjects. If you want to use
>>>     TTS then you need the "speak" app, not the "say" app:
>>>     http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_speak
>>>
>>>     -MC
>>>
>>>     On Wed, Jun 22, 2011 at 12:53 PM, Wes <wes-fs at 499x.com
>>>     <mailto:wes-fs at 499x.com>> wrote:
>>>
>>>         I'm trying to convert an example from a dialplan:
>>>
>>>         http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits#Examples
>>>
>>>         <extension name="play_and_get_digits with say">
>>>         <condition field="destination_number" expression="^(6500)$">
>>>         <action application="play_and_get_digits" data="1 1 1 3000 #
>>>         say:'press one for technical support' silence_stream://250
>>>         res \d+" />
>>>         </condition>
>>>         </extension>
>>>
>>>         into a lua statement like this:
>>>
>>>          digits = session:playAndGetDigits(1, 1, 1, 3000, "#",
>>>         "say:'press one for technicial support'
>>>         ", "say:'invalid entry'", "\\d+")
>>>
>>>         but I'm having trouble with the quoting.  The log says:
>>>
>>>         2011-06-22 14:40:52.614654 [ERR] switch_ivr_play_say.c:1144
>>>         Invalid Args
>>>
>>>         I can't find any lua examples that use both playAndGetDigits
>>>         and "say"
>>>
>>>         any suggestions?
>>>
>>>
>>>         _______________________________________________
>>>         Join us at ClueCon 2011, Aug 9-11, Chicago
>>>         http://www.cluecon.com 877-7-4ACLUE
>>>
>>>         FreeSWITCH-users mailing list
>>>         FreeSWITCH-users at lists.freeswitch.org
>>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>         http://www.freeswitch.org
>>>
>>>
>>>
>>>     _______________________________________________
>>>     Join us at ClueCon 2011, Aug 9-11, Chicago
>>>     http://www.cluecon.com  877-7-4ACLUE
>>>
>>>     FreeSWITCH-users mailing list
>>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>
>>
>>     _______________________________________________
>>     Join us at ClueCon 2011, Aug 9-11, Chicago
>>     http://www.cluecon.com  877-7-4ACLUE
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>     _______________________________________________
>     Join us at ClueCon 2011, Aug 9-11, Chicago
>     http://www.cluecon.com 877-7-4ACLUE
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From msc at freeswitch.org  Fri Jun 24 20:02:58 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 24 Jun 2011 09:02:58 -0700
Subject: [Freeswitch-users] how to use session:playAndGetDigits with the
 "say" command in lua script?
In-Reply-To: <4E04B3F1.7020201@499x.com>
References: <4E02484F.90907@499x.com>
	<BANLkTikXmE0V6yxfM-HMUQ0FKOrY=K=GOw@mail.gmail.com>
	<4E025B13.90701@499x.com> <4E02681A.2070703@499x.com>
	<BANLkTi=RHa1W7qxXbQ4MTxhN22Vds97zyA@mail.gmail.com>
	<4E04B3F1.7020201@499x.com>
Message-ID: <BANLkTimx23iyhi6kpZ5Xp22R7NsfZsuQgA@mail.gmail.com>

Go look in conf/lang/en/en.xml. Near the top of that file is a place to
specify the default TTS engine and TTS voice for your language. Try setting
those to "flite" and "kal" or whatever voice you are using.
-MC

On Fri, Jun 24, 2011 at 8:57 AM, Wes <wes-fs at 499x.com> wrote:

> **
> Thanks!... I took a look, and got one working like this:
>
> <include> <!--This line will be ignored it's here to validate the xml and
> is optional -->
>         <macro name="press_x_to_continue">
>                 <input pattern="^(\d+)$">
>                         <match>
>                                 <action function="play-file"
> data="voicemail/vm-press.wav"/>
>                                 <action function="say" data="$1"
> method="pronounced" type="items"/>
>                                 <action function="play-file"
> data="voicemail/vm-continue.wav"/>
>                                 <action function="speak-text" data="yo, you
> must press $1 to continue, do it now"/>
>                         </match>
>                 </input>
>         </macro>
> </include>
>
> but when it got to the speak-text function, it gave this error:
>
> 2011-06-24 10:54:40.684627 [ERR] switch_core_speech.c:61 Invalid speech
> module [cepstral]!
> 2011-06-24 10:54:40.684627 [ERR] switch_ivr_play_say.c:2377 Invalid TTS
> module!
>
> I have the flite engine active, so how do I configure speak-text to try
> using the flite engine instead of cepstral?
>
>
>
> On 6/22/2011 6:42 PM, Michael Collins wrote:
>
> Have you researched the subject of "phrase macros"? That may be your golden
> ticket. If you have the FS book then look at the latter half of ch 6.
> Otherwise check out these resources:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_phrase
> conf/lang/en/vm/sounds.xml
> conf/lang/en/vm/tts.xml
>
>  The voicemail system uses phrase macros a ton, and effectively, I might
> add. In short, phrase macros let you piece together sound prompts, calls to
> the say app, calls to the speak app, calls to the sleep app, etc. I think
> you would be most interested in the "speak-text" action.
>
>  -MC
>
>
> On Wed, Jun 22, 2011 at 3:09 PM, Wes <wes-fs at 499x.com> wrote:
>
>>  or maybe we just need a new version of this method called
>> "speakAndGetDigits" that specifically takes a couple of string parameters
>> for the words to speak, instead of taking the location of the wav files...
>>
>>
>> On 6/22/2011 4:13 PM, Wes wrote:
>>
>> Sorry, I was thrown off by the dialplan example... I'm trying to use
>> speak... I changed my example to try the speak command and I still have the
>> same issue with the quotes.
>>
>> Is it possible to send in a "speak" phrase as the phrase to use in the
>> PAGD command?  Or do I have to record a wav file and pass the name of the
>> wave file in for that parameter?
>>
>> On 6/22/2011 3:51 PM, Michael Collins wrote:
>>
>> Can you confirm: are you trying to use TTS or the say engine? These are
>> two completely different subjects. If you want to use TTS then you need the
>> "speak" app, not the "say" app:
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_speak
>>
>>  -MC
>>
>> On Wed, Jun 22, 2011 at 12:53 PM, Wes <wes-fs at 499x.com> wrote:
>>
>>> I'm trying to convert an example from a dialplan:
>>>
>>>
>>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits#Examples
>>>
>>> <extension name="play_and_get_digits with say">
>>>   <condition field="destination_number" expression="^(6500)$">
>>>     <action application="play_and_get_digits" data="1 1 1 3000 #
>>> say:'press one for technical support' silence_stream://250 res \d+" />
>>>   </condition>
>>> </extension>
>>>
>>> into a lua statement like this:
>>>
>>>  digits = session:playAndGetDigits(1, 1, 1, 3000, "#", "say:'press one
>>> for technicial support'
>>> ", "say:'invalid entry'", "\\d+")
>>>
>>> but I'm having trouble with the quoting.  The log says:
>>>
>>> 2011-06-22 14:40:52.614654 [ERR] switch_ivr_play_say.c:1144 Invalid Args
>>>
>>> I can't find any lua examples that use both playAndGetDigits and "say"
>>>
>>> any suggestions?
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From philippe at ppmt.org  Fri Jun 24 20:22:28 2011
From: philippe at ppmt.org (Philippe Le Toquin)
Date: Fri, 24 Jun 2011 12:22:28 -0400
Subject: [Freeswitch-users] FreeSWITCH-users@lists.freeswitch.org
In-Reply-To: <BANLkTimUx5O1Je+x7Y+46ytVPkPWk+LUhQ@mail.gmail.com>
References: <1308751418.2780.11.camel@DELL>
	<BANLkTimUx5O1Je+x7Y+46ytVPkPWk+LUhQ@mail.gmail.com>
Message-ID: <4E04B9C4.6070804@ppmt.org>

I think what he meant is that when you send an email to the list you
don't receive a copy of it
from the mailing list

It is like that for me at least. I know other receive them as they get
answered but it can be confusing

I probably won't see this actually :)

On 11-06-24 02:48 AM, Michael Collins wrote:
> I'm seeing quite a few:
>
>     * [Freeswitch-users] One way audio with Localphone .com
>       <http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073570.html> 
>       /Henk Oegema /
>     * [Freeswitch-users] One way audio with Localphone .com
>       <http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073584.html> 
>       /Henk Oegema /
>     * [Freeswitch-users] Warning - SIP auth challenge
>       <http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073587.html> 
>       /Henk Oegema /
>     * [Freeswitch-users] [Fwd: Re: Warning - SIP auth challenge]
>       <http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073589.html> 
>       /Henk Oegema /
>     * [Freeswitch-users] [Fwd: Asterisk as gateway for FS]
>       <http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073607.html> 
>       /Henk Oegema /
>     * [Freeswitch-users] No more sessions allowed at this time.
>       <http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073768.html> 
>       /Henk Oegema /
>     * [Freeswitch-users] Call internal, otherwise mobile, not working
>       <http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073814.html> 
>       /Henk Oegema /
>     * [Freeswitch-users] Call internal, otherwise mobile, not working
>       <http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073818.html> 
>       /Henk Oegema /
>     * [Freeswitch-users] FreeSWITCH-users at lists.freeswitch.org
>       <http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073848.html> 
>       /Henk Oegema /
>
> -MC
>
> On Wed, Jun 22, 2011 at 7:03 AM, Henk Oegema <henk at oegema.com
> <mailto:henk at oegema.com>> wrote:
>
>     How come I don't see my mails in the FreeSWITCH-Users Archives?
>
>
>     Rgds.
>     Henk
>
>     _______________________________________________
>     Join us at ClueCon 2011, Aug 9-11, Chicago
>     http://www.cluecon.com 877-7-4ACLUE
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From msc at freeswitch.org  Fri Jun 24 21:13:54 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 24 Jun 2011 10:13:54 -0700
Subject: [Freeswitch-users] FreeSWITCH-users@lists.freeswitch.org
In-Reply-To: <4E04B9C4.6070804@ppmt.org>
References: <1308751418.2780.11.camel@DELL>
	<BANLkTimUx5O1Je+x7Y+46ytVPkPWk+LUhQ@mail.gmail.com>
	<4E04B9C4.6070804@ppmt.org>
Message-ID: <BANLkTind0s9iQSWiJiTcvj3Z7=+hk6eYxQ@mail.gmail.com>

Yes, I don't see my own transmissions show up in my inbox, but I'm pretty
sure that's by design.
-MC

On Fri, Jun 24, 2011 at 9:22 AM, Philippe Le Toquin <philippe at ppmt.org>wrote:

> **
> I think what he meant is that when you send an email to the list you don't
> receive a copy of it
> from the mailing list
>
> It is like that for me at least. I know other receive them as they get
> answered but it can be confusing
>
> I probably won't see this actually :)
>
>
> On 11-06-24 02:48 AM, Michael Collins wrote:
>
> I'm seeing quite a few:
>
>
>    - [Freeswitch-users] One way audio with Localphone .com
>    <http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073570.html>
>      *Henk Oegema *
>    - [Freeswitch-users] One way audio with Localphone .com
>    <http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073584.html>
>      *Henk Oegema *
>    - [Freeswitch-users] Warning - SIP auth challenge
>    <http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073587.html>
>      *Henk Oegema *
>    - [Freeswitch-users] [Fwd: Re: Warning - SIP auth challenge]
>    <http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073589.html>
>      *Henk Oegema *
>    - [Freeswitch-users] [Fwd: Asterisk as gateway for FS]
>    <http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073607.html>
>      *Henk Oegema *
>    - [Freeswitch-users] No more sessions allowed at this time.
>    <http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073768.html>
>      *Henk Oegema *
>    - [Freeswitch-users] Call internal, otherwise mobile, not working
>    <http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073814.html>
>      *Henk Oegema *
>    - [Freeswitch-users] Call internal, otherwise mobile, not working
>    <http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073818.html>
>      *Henk Oegema *
>    - [Freeswitch-users] FreeSWITCH-users at lists.freeswitch.org
>    <http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073848.html>
>      *Henk Oegema *
>
> -MC
>
> On Wed, Jun 22, 2011 at 7:03 AM, Henk Oegema <henk at oegema.com> wrote:
>
>>  How come I don't see my mails in the FreeSWITCH-Users Archives?
>>
>>
>> Rgds.
>>  Henk
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vaad.fabi at gmail.com  Fri Jun 24 22:15:35 2011
From: vaad.fabi at gmail.com (vaad.fabi at gmail.com)
Date: Fri, 24 Jun 2011 21:15:35 +0300
Subject: [Freeswitch-users] mod_rtmp client API
In-Reply-To: <BANLkTimG0gN8zQzBA=pkQqDq4Q5FdYCguw@mail.gmail.com>
References: <801B6B74CA73484F96B9BBD9BE125561@e1705>	<BANLkTikmOfC7ot12-sGcO3aZ7+bH_cyGCQ@mail.gmail.com>	<4E03603B.6020405@livecall.com>	<BANLkTikAxr_SOnAAE8rbPkdhLDmXhn7cLA@mail.gmail.com>
	<BANLkTimG0gN8zQzBA=pkQqDq4Q5FdYCguw@mail.gmail.com>
Message-ID: <4E04D447.5010804@gmail.com>

  +1

On 06/23/2011 08:41 PM, Philippe Sultan wrote:
> I'm very interested too. We have a Red5 server here and some AS3 audio
> apps we'd love to test with mod_rtmp.
>
> Thanks,
>
> Philippe
>
> On Thu, Jun 23, 2011 at 7:25 PM, Anthony Minessale
> <anthony.minessale at gmail.com>  wrote:
>> lets open a new feature JIRA on it and assemble a list of participants
>> and we'll discuss it at next Wednesday public meeting, this thread and
>> IRC.
>>
>>
>> On Thu, Jun 23, 2011 at 10:48 AM, Jack<jack at livecall.com>  wrote:
>>> I am interested....  It looks like a clean, simple way to connect to
>>> Freeswitch.
>>> Nice Job!
>>> Jack
>>>
>>> On 6/22/2011 9:19 AM, Anthony Minessale wrote:
>>>> Actually......
>>>>
>>>> We have something we are working on that we are considering releasing
>>>> if we can build a community base effort around it.
>>>> When I get a sign of enough interest we'll go from there.
>>>>
>>>> It would be nice to do some presentations at ClueCon about it.
>>>>
>>>>
>>>> On Wed, Jun 22, 2011 at 10:53 AM, Madovsky<infos at madovsky.org>    wrote:
>>>>> if I type "actionscript mod_rtmp freeswitch" on google
>>>>> nothing that helps a common developer to work on mod_rtmp
>>>>> ....
>>>>>
>>>>>
>>>>> ----- Original Message -----
>>>>> From: Madovsky
>>>>> To: FreeSWITCH Users Help
>>>>> Sent: Wednesday, June 22, 2011 11:20 AM
>>>>> Subject: Re: [Freeswitch-users] mod_rtmp client API
>>>>> Mike,
>>>>>
>>>>> I didn't ask tools, but to know at least which function it needs to call to
>>>>> the server
>>>>> to make it work ! :)
>>>>>
>>>>> ----- Original Message -----
>>>>> From: Michael Collins
>>>>> To: FreeSWITCH Users Help
>>>>> Sent: Wednesday, June 22, 2011 11:11 AM
>>>>> Subject: Re: [Freeswitch-users] mod_rtmp client API
>>>>> No, it's a DIY project. We provided the server, you provide the client. :D
>>>>> Google around and you will find various tools and pieces to start building
>>>>> your own.
>>>>> -MC
>>>>>
>>>>> On Wed, Jun 22, 2011 at 7:52 AM, Madovsky<infos at madovsky.org>    wrote:
>>>>>> Is there any client side API to use mod_rtmp ?
>>>>>>
>>>>>> Thanks
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>> ________________________________
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>


-- 
Vadim F.



From wes-fs at 499x.com  Fri Jun 24 23:39:00 2011
From: wes-fs at 499x.com (Wes)
Date: Fri, 24 Jun 2011 14:39:00 -0500
Subject: [Freeswitch-users] how to use session:playAndGetDigits with the
 "say" command in lua script?
In-Reply-To: <BANLkTimx23iyhi6kpZ5Xp22R7NsfZsuQgA@mail.gmail.com>
References: <4E02484F.90907@499x.com>	<BANLkTikXmE0V6yxfM-HMUQ0FKOrY=K=GOw@mail.gmail.com>	<4E025B13.90701@499x.com>
	<4E02681A.2070703@499x.com>	<BANLkTi=RHa1W7qxXbQ4MTxhN22Vds97zyA@mail.gmail.com>	<4E04B3F1.7020201@499x.com>
	<BANLkTimx23iyhi6kpZ5Xp22R7NsfZsuQgA@mail.gmail.com>
Message-ID: <4E04E7D4.4090004@499x.com>

worked great, thank you

On 6/24/2011 11:02 AM, Michael Collins wrote:
> Go look in conf/lang/en/en.xml. Near the top of that file is a place 
> to specify the default TTS engine and TTS voice for your language. Try 
> setting those to "flite" and "kal" or whatever voice you are using.
> -MC
>
> On Fri, Jun 24, 2011 at 8:57 AM, Wes <wes-fs at 499x.com 
> <mailto:wes-fs at 499x.com>> wrote:
>
>     Thanks!... I took a look, and got one working like this:
>
>     <include> <!--This line will be ignored it's here to validate the
>     xml and is optional -->
>     <macro name="press_x_to_continue">
>     <input pattern="^(\d+)$">
>     <match>
>     <action function="play-file" data="voicemail/vm-press.wav"/>
>     <action function="say" data="$1" method="pronounced" type="items"/>
>     <action function="play-file" data="voicemail/vm-continue.wav"/>
>     <action function="speak-text" data="yo, you must press $1 to
>     continue, do it now"/>
>     </match>
>     </input>
>     </macro>
>     </include>
>
>     but when it got to the speak-text function, it gave this error:
>
>     2011-06-24 10:54:40.684627 [ERR] switch_core_speech.c:61 Invalid
>     speech module [cepstral]!
>     2011-06-24 10:54:40.684627 [ERR] switch_ivr_play_say.c:2377
>     Invalid TTS module!
>
>     I have the flite engine active, so how do I configure speak-text
>     to try using the flite engine instead of cepstral?
>
>
>
>     On 6/22/2011 6:42 PM, Michael Collins wrote:
>>     Have you researched the subject of "phrase macros"? That may be
>>     your golden ticket. If you have the FS book then look at the
>>     latter half of ch 6. Otherwise check out these resources:
>>     http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_phrase
>>     conf/lang/en/vm/sounds.xml
>>     conf/lang/en/vm/tts.xml
>>
>>     The voicemail system uses phrase macros a ton, and effectively, I
>>     might add. In short, phrase macros let you piece together sound
>>     prompts, calls to the say app, calls to the speak app, calls to
>>     the sleep app, etc. I think you would be most interested in the
>>     "speak-text" action.
>>
>>     -MC
>>
>>
>>     On Wed, Jun 22, 2011 at 3:09 PM, Wes <wes-fs at 499x.com
>>     <mailto:wes-fs at 499x.com>> wrote:
>>
>>         or maybe we just need a new version of this method called
>>         "speakAndGetDigits" that specifically takes a couple of
>>         string parameters for the words to speak, instead of taking
>>         the location of the wav files...
>>
>>
>>         On 6/22/2011 4:13 PM, Wes wrote:
>>>         Sorry, I was thrown off by the dialplan example... I'm
>>>         trying to use speak... I changed my example to try the speak
>>>         command and I still have the same issue with the quotes.
>>>
>>>         Is it possible to send in a "speak" phrase as the phrase to
>>>         use in the PAGD command?  Or do I have to record a wav file
>>>         and pass the name of the wave file in for that parameter?
>>>
>>>         On 6/22/2011 3:51 PM, Michael Collins wrote:
>>>>         Can you confirm: are you trying to use TTS or the say
>>>>         engine? These are two completely different subjects. If you
>>>>         want to use TTS then you need the "speak" app, not the
>>>>         "say" app:
>>>>         http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_speak
>>>>
>>>>         -MC
>>>>
>>>>         On Wed, Jun 22, 2011 at 12:53 PM, Wes <wes-fs at 499x.com
>>>>         <mailto:wes-fs at 499x.com>> wrote:
>>>>
>>>>             I'm trying to convert an example from a dialplan:
>>>>
>>>>             http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits#Examples
>>>>
>>>>             <extension name="play_and_get_digits with say">
>>>>             <condition field="destination_number"
>>>>             expression="^(6500)$">
>>>>             <action application="play_and_get_digits" data="1 1 1
>>>>             3000 # say:'press one for technical support'
>>>>             silence_stream://250 res \d+" />
>>>>             </condition>
>>>>             </extension>
>>>>
>>>>             into a lua statement like this:
>>>>
>>>>              digits = session:playAndGetDigits(1, 1, 1, 3000, "#",
>>>>             "say:'press one for technicial support'
>>>>             ", "say:'invalid entry'", "\\d+")
>>>>
>>>>             but I'm having trouble with the quoting.  The log says:
>>>>
>>>>             2011-06-22 14:40:52.614654 [ERR]
>>>>             switch_ivr_play_say.c:1144 Invalid Args
>>>>
>>>>             I can't find any lua examples that use both
>>>>             playAndGetDigits and "say"
>>>>
>>>>             any suggestions?
>>>>
>>>>
>>>>             _______________________________________________
>>>>             Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>             http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>>             FreeSWITCH-users mailing list
>>>>             FreeSWITCH-users at lists.freeswitch.org
>>>>             <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>             http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>         _______________________________________________
>>>>         Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>         http://www.cluecon.com  877-7-4ACLUE
>>>>
>>>>         FreeSWITCH-users mailing list
>>>>         FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>         http://www.freeswitch.org
>>>
>>>
>>>         _______________________________________________
>>>         Join us at ClueCon 2011, Aug 9-11, Chicago
>>>         http://www.cluecon.com  877-7-4ACLUE
>>>
>>>         FreeSWITCH-users mailing list
>>>         FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>         http://www.freeswitch.org
>>
>>         _______________________________________________
>>         Join us at ClueCon 2011, Aug 9-11, Chicago
>>         http://www.cluecon.com 877-7-4ACLUE
>>
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>     _______________________________________________
>>     Join us at ClueCon 2011, Aug 9-11, Chicago
>>     http://www.cluecon.com  877-7-4ACLUE
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>     _______________________________________________
>     Join us at ClueCon 2011, Aug 9-11, Chicago
>     http://www.cluecon.com 877-7-4ACLUE
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From wes-fs at 499x.com  Fri Jun 24 23:51:03 2011
From: wes-fs at 499x.com (Wes)
Date: Fri, 24 Jun 2011 14:51:03 -0500
Subject: [Freeswitch-users] How to deal with user pressing extra keys?
	flush_dtmf? session::flushDigits?
Message-ID: <4E04EAA7.2050305@499x.com>

I'm writing an ivr script in lua, and using playAndGetDigits in a couple 
different places.  Both places just take a single digit to continue on. 
If the user presses 2 digits, it seems as if the second digit is being 
queued and it's being processed in the next call to playAndGetDigits.  
Often the extra digit is invalid, so when it tells the user to press a 
certain key, it follows immediately with the message "invalid key", due 
to the extra key being in the "queue"...

is this anything I can protect against?   I found a flush_dtmf, but it 
is not very documented:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_flush_dtmf

and I also found this:
http://wiki.freeswitch.org/wiki/Session_flushDigits

but calling it before and after playAndGetDigits didn't *seem* to 
help... should it?

script here:
http://pastebin.freeswitch.org/16584





From bmoore at statirasystems.com  Sat Jun 25 00:00:57 2011
From: bmoore at statirasystems.com (William Moore)
Date: Fri, 24 Jun 2011 16:00:57 -0400
Subject: [Freeswitch-users] CRM Integration or Softphone with CRM integration
Message-ID: <4E04ECF9.5080709@statirasystems.com>

I tried searching through the archives. It turn up nothing.

Were can I find information on integrating freeswitch in to SugarCRM? 
And/Or does anyone know of a soft phone that could help with this?

What I am looking for is something that will pop up or give a choice for 
popping up a SugarCRM account based on the caller ID.

Thanks,

-- 
William J. Moore
Statira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com



From msc at freeswitch.org  Sat Jun 25 00:10:28 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 24 Jun 2011 13:10:28 -0700
Subject: [Freeswitch-users] CRM Integration or Softphone with CRM
	integration
In-Reply-To: <4E04ECF9.5080709@statirasystems.com>
References: <4E04ECF9.5080709@statirasystems.com>
Message-ID: <BANLkTi=T9s1ouYrVVoMy+Bd57=R-Fs_2kQ@mail.gmail.com>

Excellent question. I'm not familiar with anyone who has done a FS-specific
implementation for SugarCRM (or any other CRM). I am interested in knowing
more if anyone has built such a solution. That would be a good candidate for
the weekly conference calls.

Thanks,
MC

On Fri, Jun 24, 2011 at 1:00 PM, William Moore <bmoore at statirasystems.com>wrote:

> I tried searching through the archives. It turn up nothing.
>
> Were can I find information on integrating freeswitch in to SugarCRM?
> And/Or does anyone know of a soft phone that could help with this?
>
> What I am looking for is something that will pop up or give a choice for
> popping up a SugarCRM account based on the caller ID.
>
> Thanks,
>
> --
> William J. Moore
> Statira Systems
> 611 Caroline St
> Fredericksburg, VA
> http://www.statirasystems.com
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gmaruzz at gmail.com  Sat Jun 25 00:12:42 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 24 Jun 2011 22:12:42 +0200
Subject: [Freeswitch-users] CRM Integration or Softphone with CRM
	integration
In-Reply-To: <BANLkTi=T9s1ouYrVVoMy+Bd57=R-Fs_2kQ@mail.gmail.com>
References: <4E04ECF9.5080709@statirasystems.com>
	<BANLkTi=T9s1ouYrVVoMy+Bd57=R-Fs_2kQ@mail.gmail.com>
Message-ID: <BANLkTin9OLX1J_49078Cj-PeSGSPEaX8BQ@mail.gmail.com>

OpenACD ?

On 6/24/11, Michael Collins <msc at freeswitch.org> wrote:
> Excellent question. I'm not familiar with anyone who has done a FS-specific
> implementation for SugarCRM (or any other CRM). I am interested in knowing
> more if anyone has built such a solution. That would be a good candidate for
> the weekly conference calls.
>
> Thanks,
> MC
>
> On Fri, Jun 24, 2011 at 1:00 PM, William Moore
> <bmoore at statirasystems.com>wrote:
>
>> I tried searching through the archives. It turn up nothing.
>>
>> Were can I find information on integrating freeswitch in to SugarCRM?
>> And/Or does anyone know of a soft phone that could help with this?
>>
>> What I am looking for is something that will pop up or give a choice for
>> popping up a SugarCRM account based on the caller ID.
>>
>> Thanks,
>>
>> --
>> William J. Moore
>> Statira Systems
>> 611 Caroline St
>> Fredericksburg, VA
>> http://www.statirasystems.com
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From anthony.minessale at gmail.com  Sat Jun 25 00:17:14 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 24 Jun 2011 15:17:14 -0500
Subject: [Freeswitch-users] How to deal with user pressing extra keys?
 flush_dtmf? session::flushDigits?
In-Reply-To: <4E04EAA7.2050305@499x.com>
References: <4E04EAA7.2050305@499x.com>
Message-ID: <BANLkTi=aaaCs=nXzDU9R_9C1-OLZMFZhiA@mail.gmail.com>

session:flushDigits() right before you collect DTMF and you will never
get any old dtmf


On Fri, Jun 24, 2011 at 2:51 PM, Wes <wes-fs at 499x.com> wrote:
> I'm writing an ivr script in lua, and using playAndGetDigits in a couple
> different places. ?Both places just take a single digit to continue on.
> If the user presses 2 digits, it seems as if the second digit is being
> queued and it's being processed in the next call to playAndGetDigits.
> Often the extra digit is invalid, so when it tells the user to press a
> certain key, it follows immediately with the message "invalid key", due
> to the extra key being in the "queue"...
>
> is this anything I can protect against? ? I found a flush_dtmf, but it
> is not very documented:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_flush_dtmf
>
> and I also found this:
> http://wiki.freeswitch.org/wiki/Session_flushDigits
>
> but calling it before and after playAndGetDigits didn't *seem* to
> help... should it?
>
> script here:
> http://pastebin.freeswitch.org/16584
>
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Sat Jun 25 00:22:05 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 24 Jun 2011 15:22:05 -0500
Subject: [Freeswitch-users] CRM Integration or Softphone with CRM
	integration
In-Reply-To: <4E04ECF9.5080709@statirasystems.com>
References: <4E04ECF9.5080709@statirasystems.com>
Message-ID: <BANLkTinBK0=Z1OM0EZuXsF9mPtnQLn+TUg@mail.gmail.com>

write an app with libesl to connect to FS and listen for events and
generate the os call to open the link or make a web based one with
your favorite comet lib and server side ESL


On Fri, Jun 24, 2011 at 3:00 PM, William Moore
<bmoore at statirasystems.com> wrote:
> I tried searching through the archives. It turn up nothing.
>
> Were can I find information on integrating freeswitch in to SugarCRM?
> And/Or does anyone know of a soft phone that could help with this?
>
> What I am looking for is something that will pop up or give a choice for
> popping up a SugarCRM account based on the caller ID.
>
> Thanks,
>
> --
> William J. Moore
> Statira Systems
> 611 Caroline St
> Fredericksburg, VA
> http://www.statirasystems.com
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From wes-fs at 499x.com  Sat Jun 25 01:06:37 2011
From: wes-fs at 499x.com (Wes)
Date: Fri, 24 Jun 2011 16:06:37 -0500
Subject: [Freeswitch-users] How to deal with user pressing extra keys?
 flush_dtmf? session::flushDigits?
In-Reply-To: <BANLkTi=aaaCs=nXzDU9R_9C1-OLZMFZhiA@mail.gmail.com>
References: <4E04EAA7.2050305@499x.com>
	<BANLkTi=aaaCs=nXzDU9R_9C1-OLZMFZhiA@mail.gmail.com>
Message-ID: <4E04FC5D.4010900@499x.com>

I put flushDigits right before playAndGetDigits, but it is still 
queueing up keypresses... if I type more than one key, weird things 
start happening in my loop.  (the one that prompts for 1 - play back 
recording, 2=submit recording, 3 = rerecord)... sometimes I get the 
message "invalid response" followed a few seconds by "your recording has 
been submitted", and I've not typed anything in between.

incidentally, if I press a digit while playAndGetDigits is still 
speaking, it takes a few seconds for it to stop talking and act on the 
keypress... ... it just doesn't seem very responsive.... which is why 
I'm testing extra keypresses... because if users don't immediately get a 
response, they tend to press the key again.

script here:
http://pastebin.freeswitch.org/16584



On 6/24/2011 3:17 PM, Anthony Minessale wrote:
> session:flushDigits() right before you collect DTMF and you will never
> get any old dtmf
>
>
> On Fri, Jun 24, 2011 at 2:51 PM, Wes<wes-fs at 499x.com>  wrote:
>> I'm writing an ivr script in lua, and using playAndGetDigits in a couple
>> different places.  Both places just take a single digit to continue on.
>> If the user presses 2 digits, it seems as if the second digit is being
>> queued and it's being processed in the next call to playAndGetDigits.
>> Often the extra digit is invalid, so when it tells the user to press a
>> certain key, it follows immediately with the message "invalid key", due
>> to the extra key being in the "queue"...
>>
>> is this anything I can protect against?   I found a flush_dtmf, but it
>> is not very documented:
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_flush_dtmf
>>
>> and I also found this:
>> http://wiki.freeswitch.org/wiki/Session_flushDigits
>>
>> but calling it before and after playAndGetDigits didn't *seem* to
>> help... should it?
>>
>> script here:
>> http://pastebin.freeswitch.org/16584
>>
>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>


From wes-fs at 499x.com  Sat Jun 25 01:26:06 2011
From: wes-fs at 499x.com (Wes)
Date: Fri, 24 Jun 2011 16:26:06 -0500
Subject: [Freeswitch-users] .wav vs .gsm file sizes for recording calls.
Message-ID: <4E0500EE.9040506@499x.com>

In my tests, if I record a call in .wav format, a 10 second file is 
about 177,000 bytes, while a 10 second .gsm file is 17,000 bytes.

I  then used sox to convert the .gsm file to a .wav file, and it stayed 
at around 17,000 bytes.  So, is the default recording format for .wav 
using a higher sample rate? vs the default conversion format for the sox 
tool?

checking the file type using "file" I see that the larger one is:
RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz

and the wav created by sox via the default conversion from .gsm is:
RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz

So apparently the larger wav file is 16 bit... how are these recording 
parameters controlled?  Can I set it to record directly into the smaller 
wav format? Or will I have to run sox on every file...


From wes-fs at 499x.com  Sat Jun 25 01:37:39 2011
From: wes-fs at 499x.com (Wes)
Date: Fri, 24 Jun 2011 16:37:39 -0500
Subject: [Freeswitch-users] getting rid of hanging sessions?
Message-ID: <4E0503A3.5040909@499x.com>

in my testing, I'm getting some sessions that are, hung, I guess.

Status shows:

UP 0 years, 0 days, 2 hours, 7 minutes, 37 seconds, 752 milliseconds, 
720 microseconds
23 session(s) since startup
5 session(s) 0/30
1000 session(s) max
min idle cpu 0.00/0.00


if I do a shutdown, it waits on them for a while, then eventually shuts 
down.

Is there any way to "clean" these up periodically?  Or, what am I doing 
to leave them out there?

thanks!


From david.ponzone at ipeva.fr  Sat Jun 25 01:41:51 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 24 Jun 2011 23:41:51 +0200
Subject: [Freeswitch-users] .wav vs .gsm file sizes for recording calls.
In-Reply-To: <4E0500EE.9040506@499x.com>
References: <4E0500EE.9040506@499x.com>
Message-ID: <E813864E-334B-4779-A000-DB50333ED717@ipeva.fr>

Wes,

it's because you are confusing 2 things.
GSM is a codec, so it's a sound format.
WAVE is a kind of container (like MKV) that can contain various codecs.
In your case, you probably did a mistake using sox (it's not an easy tool), so you ended up with a wav which is still GSM audio inside.

You have to tell sox which exact output format you want.
AFAIR, you have to use the -s option (for signed linear).

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

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

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




Le 24/06/2011 ? 23:26, Wes a ?crit :

> In my tests, if I record a call in .wav format, a 10 second file is 
> about 177,000 bytes, while a 10 second .gsm file is 17,000 bytes.
> 
> I  then used sox to convert the .gsm file to a .wav file, and it stayed 
> at around 17,000 bytes.  So, is the default recording format for .wav 
> using a higher sample rate? vs the default conversion format for the sox 
> tool?
> 
> checking the file type using "file" I see that the larger one is:
> RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
> 
> and the wav created by sox via the default conversion from .gsm is:
> RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz
> 
> So apparently the larger wav file is 16 bit... how are these recording 
> parameters controlled?  Can I set it to record directly into the smaller 
> wav format? Or will I have to run sox on every file...
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jan.berger at video24.no  Sat Jun 25 01:46:00 2011
From: jan.berger at video24.no (Jan Berger)
Date: Fri, 24 Jun 2011 23:46:00 +0200
Subject: [Freeswitch-users] .wav vs .gsm file sizes for recording calls.
In-Reply-To: <4E0500EE.9040506@499x.com>
References: <4E0500EE.9040506@499x.com>
Message-ID: <A9B6F94D5E28431C9B31134FBDF47738@dell9400>

Wav is just a standard file format, the size is decided by the contents.

The first format is PCM which is 8000 bytes samples per sec
The second is GSM which is compressed voice - so in short you changed the
file format, but is was and still is GSM inside.

To convert from PCM to GSM you need an application that can do so - WavePad
etc. can be recommended.

Jan

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Wes
Sent: 24. juni 2011 23:26
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] .wav vs .gsm file sizes for recording calls.

In my tests, if I record a call in .wav format, a 10 second file is 
about 177,000 bytes, while a 10 second .gsm file is 17,000 bytes.

I  then used sox to convert the .gsm file to a .wav file, and it stayed 
at around 17,000 bytes.  So, is the default recording format for .wav 
using a higher sample rate? vs the default conversion format for the sox 
tool?

checking the file type using "file" I see that the larger one is:
RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz

and the wav created by sox via the default conversion from .gsm is:
RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz

So apparently the larger wav file is 16 bit... how are these recording 
parameters controlled?  Can I set it to record directly into the smaller 
wav format? Or will I have to run sox on every file...

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

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From jan.berger at video24.no  Sat Jun 25 01:53:51 2011
From: jan.berger at video24.no (Jan Berger)
Date: Fri, 24 Jun 2011 23:53:51 +0200
Subject: [Freeswitch-users] .wav vs .gsm file sizes for recording calls.
In-Reply-To: <E813864E-334B-4779-A000-DB50333ED717@ipeva.fr>
References: <4E0500EE.9040506@499x.com>
	<E813864E-334B-4779-A000-DB50333ED717@ipeva.fr>
Message-ID: <14F4842598F24E4EB19AB7735FE9B89F@dell9400>

A gentle warning when using GSM codec?s

 

Every codec compress voice at the cost of quality. And some codec?s like GSM
and G.729 are not a nice combination. If you use GSM to save disk space and
play it back using G.729 the resulting voice might contain some added noise
due to algorithmic incompatibility.

 

jan

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David
Ponzone
Sent: 24. juni 2011 23:42
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] .wav vs .gsm file sizes for recording calls.

 

Wes,

 

it's because you are confusing 2 things.

GSM is a codec, so it's a sound format.

WAVE is a kind of container (like MKV) that can contain various codecs.

In your case, you probably did a mistake using sox (it's not an easy tool),
so you ended up with a wav which is still GSM audio inside.

 

You have to tell sox which exact output format you want.

AFAIR, you have to use the -s option (for signed linear).

 

David Ponzone  Direction Technique

email: david.ponzone at ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

 

Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>   -   www.ipeva-studio.com
<BLOCKED::http://www.ipeva-studio.com/> 

 

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

 





 

Le 24/06/2011 ? 23:26, Wes a ?crit :





In my tests, if I record a call in .wav format, a 10 second file is 
about 177,000 bytes, while a 10 second .gsm file is 17,000 bytes.

I  then used sox to convert the .gsm file to a .wav file, and it stayed 
at around 17,000 bytes.  So, is the default recording format for .wav 
using a higher sample rate? vs the default conversion format for the sox 
tool?

checking the file type using "file" I see that the larger one is:
RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz

and the wav created by sox via the default conversion from .gsm is:
RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz

So apparently the larger wav file is 16 bit... how are these recording 
parameters controlled?  Can I set it to record directly into the smaller 
wav format? Or will I have to run sox on every file...

_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
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From d at d-man.org  Sat Jun 25 03:09:42 2011
From: d at d-man.org (Darren Schreiber)
Date: Fri, 24 Jun 2011 16:09:42 -0700
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <4E04B039.6020706@statirasystems.com>
References: <CA293ED7.2B6A6%d@d-man.org> <4E04B039.6020706@statirasystems.com>
Message-ID: <196C835FADC4D243A06AA032715AE7211E1A2988@EXVMBX020-20.exch020.serverdata.net>

So one thing we are working on is an install wizard cause most of the issues we hear about are people not understanding where to go to set things up.

The problem is that blue.box, via 3 checkboxes and/or dropdowns, can be:

-          A premise/hosted single-tenant PBX that simplifies setup by forcing a domain and auto-activating RPort and a few other items

-          A complex hosted PBX w/ multi-tenant that works based on domain names

-          A simple routing/bridging tool w/ no PBX features

That seems to be how most people are using it. I think we will make those options on startup. Also, we need to simplify ACLs.


-          Darren


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of William Moore
Sent: Friday, June 24, 2011 8:42 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Trunk Issue

Thanks for responding. I just stayed up all night learning all I could about both freeswitch and blue.box. I had a client pushing me so I was a little frustrated with it. I usually troll and learn it on my own via the code, documentation, bug lists, and so on. I was looking for a more immediate response than normal.

I do appreciate the project and all the work put forward by you and your team.

The only other software I had as much trouble learning was when I switched to Gentoo and found out how little I knew about linux.

You will hear from me again. Hopefully via contributions.


William J. Moore

Statira Systems

611 Caroline St

Fredericksburg, VA

http://www.statirasystems.com

On 06/23/2011 10:05 PM, Darren Schreiber wrote:
OK, well again, I'm happy to help, as is my team and as are the open-source folks. Sometimes we do get busy, so please just ping me, but the logs pasted by William indicate he doesn't know how to setup the authentication for his carriers/phones and since there's 82 different ways to do that, I'd need to know more info.

Hence the request to email off list, unless he wants to post his auth strategy and IPs publicly :)

--


From: David Ponzone <david.ponzone at ipeva.fr<mailto:david.ponzone at ipeva.fr>>
Reply-To: "freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>" <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Date: Thu, 23 Jun 2011 18:56:04 -0700
To: "freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>" <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] Trunk Issue

No Darren, sorry, I confused FreePBX and FusionPBX.

My bad.

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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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



Le 24/06/2011 ? 03:14, Darren Schreiber a ?crit :


David,
FreePBX v3 is now called blue.box. You're not making any sense... Is this an attempt at creating FUD?

- Darren

--


From: David Ponzone <david.ponzone at ipeva.fr<mailto:david.ponzone at ipeva.fr>>
Reply-To: "freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>" <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Date: Thu, 23 Jun 2011 12:59:16 -0700
To: "freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>" <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] Trunk Issue

I had the same feeling about blue.box.
I think there are more interesting GUIs out there, like FreePBX V3.
The weird thing is that none of them seems very active.

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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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



Le 23/06/2011 ? 19:55, William Moore a ?crit :


I give up on blue.box. They seem preoccupied by there other projects. Documentation does not explain enough. It is like describing a human by their parts only and not how it interacts. Peace out.


William J. Moore

Statira Systems

611 Caroline St

Fredericksburg, VA

http://www.statirasystems.com<http://www.statirasystems.com/>

On 06/23/2011 01:34 PM, David Ponzone wrote:
In sipinterface_2, make the following modification:

 <param name="auth-calls" value="false"/>





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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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




Le 23/06/2011 ? 19:31, David Ponzone a ?crit :


The issue is that you are expecting them to authenticate, but they won't.

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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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




Le 23/06/2011 ? 19:15, William Moore a ?crit :


Ok, I reregistered and called right away. I get the following:

  0.000000 66.227.100.20 -> 192.168.1.254 SIP/SDP Request: INVITE sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with session description

  0.000766 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying

  0.002321 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy Authentication Required

  0.021120 66.227.100.20 -> 192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2

  3.272168 192.168.1.254 -> 66.227.100.20 SIP Request: REGISTER sip:jnctn.net;transport=udp

  3.292076 66.227.100.20 -> 192.168.1.254 SIP Status: 200 OK    (1 bindings)



The call just states it is a none working number. That means something in my dial plan is incorrect. I am double checking it now.

William J. MooreStatira Systems

611 Caroline St

Fredericksburg, VA

http://www.statirasystems.com<http://www.statirasystems.com/>

On 06/23/2011 12:58 PM, William Moore wrote:
When I put it in and it reloads, blue.box removes it. Can I reference the gateway in another sip interface file or does it have to be in that one?


William J. Moore

Statira Systems

611 Caroline St

Fredericksburg, VA

http://www.statirasystems.com<http://www.statirasystems.com/>

On 06/23/2011 12:52 PM, William Moore wrote:
correction: missed a line.

  0.000000 66.227.100.20 -> 192.168.1.254 SIP/SDP Request: INVITE sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with session description
  0.000814 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
  0.002440 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy Authentication Required
  0.023173 66.227.100.20 -> 192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2



William J. Moore

Statira Systems

611 Caroline St

Fredericksburg, VA

http://www.statirasystems.com<http://www.statirasystems.com/>

On 06/23/2011 12:50 PM, William Moore wrote:
I get this with wireshark

 40.345679 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
 40.347381 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy Authentication Required
 40.367906 66.227.100.20 -> 192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2

I am looking for the expiration now..


William J. Moore

Statira Systems

611 Caroline St

Fredericksburg, VA

http://www.statirasystems.com<http://www.statirasystems.com/>

On 06/23/2011 12:25 PM, David Ponzone wrote:
Take a sip trace on your box with tcpdump or preferably, tshark:
tshark port 5080

You will then see if you receive it.

In the config you sent, I don't see anything about register expiration, or NAT keepalive, so I would really recommend you add a ping every 30 seconds to your gateway config.

If the issue is there, it's quite easy to see:
unregister the gateway
register it again
make an inbound call right away

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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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




Le 23/06/2011 ? 18:15, William Moore a ?crit :


It was the 1 and it needs to be set for e.164(+1). Blue.box makes it a little more difficult since it wants to manage the xmls. I could have set it up as a separate config file however that defeats the purpose of the GUI.

I still have the issue of incoming calls not even showing up in sofia. Junction Networks says it is going out, but my server is not responding. I'm not sure if I have the settings for incoming correct.


William J. Moore

Statira Systems

611 Caroline St

Fredericksburg, VA

http://www.statirasystems.com<http://www.statirasystems.com/>

On 06/23/2011 04:21 AM, David Ponzone wrote:
William,

If I believe the example configuraiton on Junction's web, they don't expect E164:


   <extension name="junctionnetworks.com<http://junctionnetworks.com/>>

    <condition field="destination_number" expression="^(1{0,1}\d{10})$">

      <action application="set"data="effective_caller_id_number=OUTBOUNDCALLERID"/>

      <action application="bridge" data="sofia/gateway/junctionnetworks.com/$1"/>

     </condition>

   </extension>
But, elsewhere in their knowledgebase, they say:


Numbering
The standard e.164 numbering plan (ITU<http://www.itu.int/>) is used. North American numbers are required to be prefixed with a '1'. International numbers need to be prefixed with "011".

I think they are confused about what is E164....

You should try to prefix with 1 as Michael recommended.

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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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




Le 23/06/2011 ? 08:19, Michael Collins a ?crit :


Perhaps the gateway wants the leading 1 or expects e.164 format?
-MC
On Wed, Jun 22, 2011 at 5:37 PM, Brian West <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:

484 is address incomplete means you didn't send enough digits to the gateway usually.

/b

On Jun 22, 2011, at 7:34 PM, Brad Mina wrote:


Ah, thought it was recv not send. Good to know that an invalid gateway issue
can cause this.

On Wed, Jun 22, 2011 at 5:31 PM, Michael Collins <msc at freeswitch.org<mailto:msc at freeswitch.org>> wrote:


The 484 I believe is a result of the bad gateway. In any case, the OP needs
to figure out what's up w/ that gw or no calls will be going out...

-MC



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From d at d-man.org  Sat Jun 25 03:10:45 2011
From: d at d-man.org (Darren Schreiber)
Date: Fri, 24 Jun 2011 16:10:45 -0700
Subject: [Freeswitch-users] Trunk Issue
In-Reply-To: <4E04AC28.5050402@statirasystems.com>
References: <CA293246.2B652%d@d-man.org> <4E04AC28.5050402@statirasystems.com>
Message-ID: <196C835FADC4D243A06AA032715AE7211E1A298A@EXVMBX020-20.exch020.serverdata.net>

We are working on a get started guide, too. I'll try to post that next week.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of William Moore
Sent: Friday, June 24, 2011 8:24 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Trunk Issue

LOL I know it is not difficult. The change I had an issue storing was the expire seconds. It actually stayed on the fifth try. Not sure why it would not but it is there. Actually authentication was not the issue after all. Junction sends the number with a one so I had to add it to the Number setting. Also I had the context setting on the number wrong. It kept trying to send it out the second it came in.

My concerns about the project were not about the activity of the project. It was about it's attention to the users. I tried the documentation, which is better then a lot of projects I have looked at, and it failed me. It is only about each specific thing and yet it is not that specific. To someone well versed in freeswitch, they could connect the dots, my green self could not.

Also that comments I made were because I had attempted assistance via 2600hz IRC and forum. As you can see I ended up here.




William J. Moore

Statira Systems

611 Caroline St

Fredericksburg, VA

http://www.statirasystems.com

On 06/23/2011 09:13 PM, Darren Schreiber wrote:
Hi William,
Sorry to hear that - what exactly makes you think this? We're very active on the blue.box front. There have been 3 new features and over 20 bug fixes in just the past four weeks. Can you email me off-list to get whatever issue you're having sorted out?

Also, for authentication, bluebox "replaces" your changes because you're supposed to use the GUI to change that option. Go into Connectivity / SIP Interfaces and edit the SIP port you're trying to change. Uncheck the mark named "Require Authentication". It's not very difficult...

- Darren

--


From: William Moore <bmoore at statirasystems.com<mailto:bmoore at statirasystems.com>>
Reply-To: "freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>" <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Date: Thu, 23 Jun 2011 10:55:15 -0700
To: "freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>" <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] Trunk Issue

I give up on blue.box. They seem preoccupied by there other projects. Documentation does not explain enough. It is like describing a human by their parts only and not how it interacts. Peace out.


William J. Moore

Statira Systems

611 Caroline St

Fredericksburg, VA

http://www.statirasystems.com

On 06/23/2011 01:34 PM, David Ponzone wrote:
In sipinterface_2, make the following modification:

 <param name="auth-calls" value="false"/>





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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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




Le 23/06/2011 ? 19:31, David Ponzone a ?crit :


The issue is that you are expecting them to authenticate, but they won't.

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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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




Le 23/06/2011 ? 19:15, William Moore a ?crit :


Ok, I reregistered and called right away. I get the following:

  0.000000 66.227.100.20 -> 192.168.1.254 SIP/SDP Request: INVITE sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with session description

  0.000766 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying

  0.002321 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy Authentication Required

  0.021120 66.227.100.20 -> 192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2

  3.272168 192.168.1.254 -> 66.227.100.20 SIP Request: REGISTER sip:jnctn.net;transport=udp

  3.292076 66.227.100.20 -> 192.168.1.254 SIP Status: 200 OK    (1 bindings)



The call just states it is a none working number. That means something in my dial plan is incorrect. I am double checking it now.

William J. MooreStatira Systems

611 Caroline St

Fredericksburg, VA

http://www.statirasystems.com<http://www.statirasystems.com/>

On 06/23/2011 12:58 PM, William Moore wrote:
When I put it in and it reloads, blue.box removes it. Can I reference the gateway in another sip interface file or does it have to be in that one?


William J. Moore

Statira Systems

611 Caroline St

Fredericksburg, VA

http://www.statirasystems.com<http://www.statirasystems.com/>

On 06/23/2011 12:52 PM, William Moore wrote:
correction: missed a line.

  0.000000 66.227.100.20 -> 192.168.1.254 SIP/SDP Request: INVITE sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2, with session description
  0.000814 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
  0.002440 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy Authentication Required
  0.023173 66.227.100.20 -> 192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2



William J. Moore

Statira Systems

611 Caroline St

Fredericksburg, VA

http://www.statirasystems.com<http://www.statirasystems.com/>

On 06/23/2011 12:50 PM, William Moore wrote:
I get this with wireshark

 40.345679 192.168.1.254 -> 66.227.100.20 SIP Status: 100 Trying
 40.347381 192.168.1.254 -> 66.227.100.20 SIP Status: 407 Proxy Authentication Required
 40.367906 66.227.100.20 -> 192.168.1.254 SIP Request: ACK sip:17036526678 at 173.72.143.16:5080;transport=udp;gw=trunk_2

I am looking for the expiration now..


William J. Moore

Statira Systems

611 Caroline St

Fredericksburg, VA

http://www.statirasystems.com<http://www.statirasystems.com/>

On 06/23/2011 12:25 PM, David Ponzone wrote:
Take a sip trace on your box with tcpdump or preferably, tshark:
tshark port 5080

You will then see if you receive it.

In the config you sent, I don't see anything about register expiration, or NAT keepalive, so I would really recommend you add a ping every 30 seconds to your gateway config.

If the issue is there, it's quite easy to see:
unregister the gateway
register it again
make an inbound call right away

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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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




Le 23/06/2011 ? 18:15, William Moore a ?crit :


It was the 1 and it needs to be set for e.164(+1). Blue.box makes it a little more difficult since it wants to manage the xmls. I could have set it up as a separate config file however that defeats the purpose of the GUI.

I still have the issue of incoming calls not even showing up in sofia. Junction Networks says it is going out, but my server is not responding. I'm not sure if I have the settings for incoming correct.


William J. Moore

Statira Systems

611 Caroline St

Fredericksburg, VA

http://www.statirasystems.com<http://www.statirasystems.com/>

On 06/23/2011 04:21 AM, David Ponzone wrote:
William,

If I believe the example configuraiton on Junction's web, they don't expect E164:


   <extension name="junctionnetworks.com<http://junctionnetworks.com/>>

    <condition field="destination_number" expression="^(1{0,1}\d{10})$">

      <action application="set"data="effective_caller_id_number=OUTBOUNDCALLERID"/>

      <action application="bridge" data="sofia/gateway/junctionnetworks.com/$1"/>

     </condition>

   </extension>
But, elsewhere in their knowledgebase, they say:


Numbering
The standard e.164 numbering plan (ITU<http://www.itu.int/>) is used. North American numbers are required to be prefixed with a '1'. International numbers need to be prefixed with "011".

I think they are confused about what is E164....

You should try to prefix with 1 as Michael recommended.

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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

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




Le 23/06/2011 ? 08:19, Michael Collins a ?crit :


Perhaps the gateway wants the leading 1 or expects e.164 format?
-MC
On Wed, Jun 22, 2011 at 5:37 PM, Brian West <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:

484 is address incomplete means you didn't send enough digits to the gateway usually.

/b

On Jun 22, 2011, at 7:34 PM, Brad Mina wrote:


Ah, thought it was recv not send. Good to know that an invalid gateway issue
can cause this.

On Wed, Jun 22, 2011 at 5:31 PM, Michael Collins <msc at freeswitch.org<mailto:msc at freeswitch.org>> wrote:


The 484 I believe is a result of the bad gateway. In any case, the OP needs
to figure out what's up w/ that gw or no calls will be going out...

-MC



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From msc at freeswitch.org  Sat Jun 25 03:13:54 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 24 Jun 2011 16:13:54 -0700
Subject: [Freeswitch-users] How to deal with user pressing extra keys?
 flush_dtmf? session::flushDigits?
In-Reply-To: <4E04FC5D.4010900@499x.com>
References: <4E04EAA7.2050305@499x.com>
	<BANLkTi=aaaCs=nXzDU9R_9C1-OLZMFZhiA@mail.gmail.com>
	<4E04FC5D.4010900@499x.com>
Message-ID: <BANLkTikvF7pxRqZw+_Dh=EWfmSKRqB-CwA@mail.gmail.com>

Hey, that Lua code looks awfully familiar! (I wrote some of that on a snowy
day in hotel room in Wisconsin...)

Okay, I think I see what you're trying to do. You just want to have the
person press 1 to continue. Do you want something different to happen if the
person presses a key other than 1? Just checking.

Also, I don't see any flushDigits in your script. Try adding this before
your PAGD:
session:execute('flush_dtmf');

Try again, report back, etc.
-MC

On Fri, Jun 24, 2011 at 2:06 PM, Wes <wes-fs at 499x.com> wrote:

> I put flushDigits right before playAndGetDigits, but it is still
> queueing up keypresses... if I type more than one key, weird things
> start happening in my loop.  (the one that prompts for 1 - play back
> recording, 2=submit recording, 3 = rerecord)... sometimes I get the
> message "invalid response" followed a few seconds by "your recording has
> been submitted", and I've not typed anything in between.
>
> incidentally, if I press a digit while playAndGetDigits is still
> speaking, it takes a few seconds for it to stop talking and act on the
> keypress... ... it just doesn't seem very responsive.... which is why
> I'm testing extra keypresses... because if users don't immediately get a
> response, they tend to press the key again.
>
> script here:
> http://pastebin.freeswitch.org/16584
>
>
>
> On 6/24/2011 3:17 PM, Anthony Minessale wrote:
> > session:flushDigits() right before you collect DTMF and you will never
> > get any old dtmf
> >
> >
> > On Fri, Jun 24, 2011 at 2:51 PM, Wes<wes-fs at 499x.com>  wrote:
> >> I'm writing an ivr script in lua, and using playAndGetDigits in a couple
> >> different places.  Both places just take a single digit to continue on.
> >> If the user presses 2 digits, it seems as if the second digit is being
> >> queued and it's being processed in the next call to playAndGetDigits.
> >> Often the extra digit is invalid, so when it tells the user to press a
> >> certain key, it follows immediately with the message "invalid key", due
> >> to the extra key being in the "queue"...
> >>
> >> is this anything I can protect against?   I found a flush_dtmf, but it
> >> is not very documented:
> >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_flush_dtmf
> >>
> >> and I also found this:
> >> http://wiki.freeswitch.org/wiki/Session_flushDigits
> >>
> >> but calling it before and after playAndGetDigits didn't *seem* to
> >> help... should it?
> >>
> >> script here:
> >> http://pastebin.freeswitch.org/16584
> >>
> >>
> >>
> >>
> >> _______________________________________________
> >> Join us at ClueCon 2011, Aug 9-11, Chicago
> >> http://www.cluecon.com 877-7-4ACLUE
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
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> >> http://www.freeswitch.org
> >>
> >
> >
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
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From msc at freeswitch.org  Sat Jun 25 03:15:10 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 24 Jun 2011 16:15:10 -0700
Subject: [Freeswitch-users] getting rid of hanging sessions?
In-Reply-To: <4E0503A3.5040909@499x.com>
References: <4E0503A3.5040909@499x.com>
Message-ID: <BANLkTinOLY6dv1tB7sZJSKQC4WMiRmEejA@mail.gmail.com>

The key is in knowing why those are stuck. Do a "show channels" when you see
these so-called phantom sessions, then put it on pastebin so we can take a
look.
-MC

On Fri, Jun 24, 2011 at 2:37 PM, Wes <wes-fs at 499x.com> wrote:

> in my testing, I'm getting some sessions that are, hung, I guess.
>
> Status shows:
>
> UP 0 years, 0 days, 2 hours, 7 minutes, 37 seconds, 752 milliseconds,
> 720 microseconds
> 23 session(s) since startup
> 5 session(s) 0/30
> 1000 session(s) max
> min idle cpu 0.00/0.00
>
>
> if I do a shutdown, it waits on them for a while, then eventually shuts
> down.
>
> Is there any way to "clean" these up periodically?  Or, what am I doing
> to leave them out there?
>
> thanks!
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
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> FreeSWITCH-users at lists.freeswitch.org
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From clive at lansink.co.nz  Sat Jun 25 03:26:18 2011
From: clive at lansink.co.nz (Clive Lansink)
Date: Sat, 25 Jun 2011 11:26:18 +1200
Subject: [Freeswitch-users] Recovering from database fault
Message-ID: <20110624232634.B7E4FE80E4@jlo.kiwilink.co.nz>

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From msc at freeswitch.org  Sat Jun 25 03:31:14 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 24 Jun 2011 16:31:14 -0700
Subject: [Freeswitch-users] .wav vs .gsm file sizes for recording calls.
In-Reply-To: <4E0500EE.9040506@499x.com>
References: <4E0500EE.9040506@499x.com>
Message-ID: <BANLkTimxdvjSU=gDXMLYLb17jDwamWDSbw@mail.gmail.com>

I would caution you to consider adding disk space before you try to compress
all your recordings. The 16 bit SLIN that FS normally puts in your wave
files are pretty easy to handle, whether playing back in a FS session, or
encoding for playback on some other device.

An alternative might be to use lame to convert them to MP3's or ogg/vorbis
files. If you look on the main FS conf call page you'll see I have the
weekly recordings in multiple formats. (
http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls)

Here are some stats for last Wednesday's call. Note that I record wave files
in 48kHz then use sox to downsample to 16kHz wave, then I convert that 16kHz
file into MP3 and Vorbis (in an ogg container). Here's what the results look
like:

<2831>:ls -1s conf_call_2011-06-15.*
 18736 conf_call_2011-06-15.mp3
 23044 conf_call_2011-06-15.ogg
199756 conf_call_2011-06-15.wav

<2832>:file conf_call_2011-06-15.mp3
conf_call_2011-06-15.mp3: MPEG ADTS, layer III, v2,  24 kBits, 16 kHz,
Monaural

<2833>:file conf_call_2011-06-15.ogg
conf_call_2011-06-15.ogg: Ogg data, Vorbis audio, mono, 16000 Hz, ~48000
bps, created by: Xiph.Org libVorbis I

<2834>:file conf_call_2011-06-15.wav
conf_call_2011-06-15.wav: RIFF (little-endian) data, WAVE audio, Microsoft
PCM, 16 bit, mono 16000 Hz

Note that the file sizes are in 1K blocks.

So, bottom line is this: if you have the disk space then use wave. If you
don't have disk space for wave then get some! :D If you REALLY need to use a
different format then choose something like MP3 or Vorbis for long-term
storage.

-MC

On Fri, Jun 24, 2011 at 2:26 PM, Wes <wes-fs at 499x.com> wrote:

> In my tests, if I record a call in .wav format, a 10 second file is
> about 177,000 bytes, while a 10 second .gsm file is 17,000 bytes.
>
> I  then used sox to convert the .gsm file to a .wav file, and it stayed
> at around 17,000 bytes.  So, is the default recording format for .wav
> using a higher sample rate? vs the default conversion format for the sox
> tool?
>
> checking the file type using "file" I see that the larger one is:
> RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
>
> and the wav created by sox via the default conversion from .gsm is:
> RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz
>
> So apparently the larger wav file is 16 bit... how are these recording
> parameters controlled?  Can I set it to record directly into the smaller
> wav format? Or will I have to run sox on every file...
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Sat Jun 25 03:36:42 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 24 Jun 2011 16:36:42 -0700
Subject: [Freeswitch-users] Recovering from database fault
In-Reply-To: <20110624232634.B7E4FE80E4@jlo.kiwilink.co.nz>
References: <20110624232634.B7E4FE80E4@jlo.kiwilink.co.nz>
Message-ID: <BANLkTi=AjFQsfpQJxKLSPVmvxtQ16vMPyg@mail.gmail.com>

On Fri, Jun 24, 2011 at 4:26 PM, Clive Lansink <clive at lansink.co.nz> wrote:

> Hello Michael and others.
>
> I deleted the whold db folder which in our case was ok because we don't use
> voice mail. Then when I ran Freeswitchconsole, the db folder was recreated.
> But that was when we had the problem. The phones could make calls but none
> could ring.
>

Okay, so you removed the whole db folder then restarted FS and everything
worked except for the registrations? Then when you did a complete re-install
of FS then everything, including registrations, seemed to work?

I'd recommend that you update to the latest git and watch and wait. If the
symptom returns then maybe you could collect a gcore and open a jira, unless
you're on Windows. (We have Windows guys who can guide you on what to do for
troubleshooting - I just don't happen to be one of them...)

-MC
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From clive at lansink.co.nz  Sat Jun 25 04:08:08 2011
From: clive at lansink.co.nz (Clive Lansink)
Date: Sat, 25 Jun 2011 12:08:08 +1200
Subject: [Freeswitch-users] Recovering from database fault
Message-ID: <20110625000823.00B8AE80E4@jlo.kiwilink.co.nz>

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From curriegrad2004 at gmail.com  Sat Jun 25 06:11:40 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Fri, 24 Jun 2011 19:11:40 -0700
Subject: [Freeswitch-users] Recovering from database fault
In-Reply-To: <20110625000823.00B8AE80E4@jlo.kiwilink.co.nz>
References: <20110625000823.00B8AE80E4@jlo.kiwilink.co.nz>
Message-ID: <BANLkTikUftc2ck=AweFLKhksjS6_3fKDUA@mail.gmail.com>

Are you running FreeSwitch as a non root user? This makes a huge
difference as I might suspect there are file permission issues that's
happening here. ie. db folder doesn't allow the fs user account to
create or write to that directory

On Fri, Jun 24, 2011 at 5:08 PM, Clive Lansink <clive at lansink.co.nz> wrote:
> Yes that's exactly right. You understand the situation correctly.
>
> I guess this situation is not going to occur often so I'll just watch to see if it ever happens again.
>
>
> Clive Lansink
> Email: Clive at Lansink.Co.NZ
> Phone: +64 9 520-4242
> Mobile: +64 21 663-999
> Fax: +64 21 789-150
>
> -----Original message-----
> From: Michael Collins <msc at freeswitch.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Recovering from database fault
> Reply-to: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Fri, 24 Jun 2011 16:36:42 -0700
>
> On Fri, Jun 24, 2011 at 4:26 PM, Clive Lansink <clive at lansink.co.nz> wrote:
>
>> Hello Michael and others.
>>
>> I deleted the whold db folder which in our case was ok because we don't use
>> voice mail. Then when I ran Freeswitchconsole, the db folder was recreated.
>> But that was when we had the problem. The phones could make calls but none
>> could ring.
>>
>
> Okay, so you removed the whole db folder then restarted FS and everything
> worked except for the registrations? Then when you did a complete re-install
> of FS then everything, including registrations, seemed to work?
>
> I'd recommend that you update to the latest git and watch and wait. If the
> symptom returns then maybe you could collect a gcore and open a jira, unless
> you're on Windows. (We have Windows guys who can guide you on what to do for
> troubleshooting - I just don't happen to be one of them...)
>
> -MC
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From bwibowo at gmail.com  Sat Jun 25 06:29:31 2011
From: bwibowo at gmail.com (mm,m)
Date: Sat, 25 Jun 2011 10:29:31 +0800 (HKT)
Subject: [Freeswitch-users] mm,m has invited you to join!
Message-ID: <857902024.1308968971335.JavaMail.tomcat@wfn-web-fail.meta4-group.com>

no text body
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From steveu at coppice.org  Sat Jun 25 07:11:42 2011
From: steveu at coppice.org (Steve Underwood)
Date: Sat, 25 Jun 2011 11:11:42 +0800
Subject: [Freeswitch-users] .wav vs .gsm file sizes for recording calls.
In-Reply-To: <14F4842598F24E4EB19AB7735FE9B89F@dell9400>
References: <4E0500EE.9040506@499x.com>	<E813864E-334B-4779-A000-DB50333ED717@ipeva.fr>
	<14F4842598F24E4EB19AB7735FE9B89F@dell9400>
Message-ID: <4E0551EE.20105@coppice.org>

Practically any conversion of a low bit rate codec to another low bit 
rate codec gives horrible results. Even low bit rate codec -> linear -> 
the same low bit rate codec can be pretty horrible, unless the two 
compression steps are synchronised.

A lot of the complaints you will find about the horrible sound from some 
codec has little to do with the codec's performance. Someone has tried 
the codec with audio that had already been compressed by some other low 
bit rate codec.

Steve

On 06/25/2011 05:53 AM, Jan Berger wrote:
>
> A gentle warning when using GSM codec?s
>
> Every codec compress voice at the cost of quality. And some codec?s 
> like GSM and G.729 are not a nice combination. If you use GSM to save 
> disk space and play it back using G.729 the resulting voice might 
> contain some added noise due to algorithmic incompatibility.
>
> jan
>
> ------------------------------------------------------------------------
>
> *From:*freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
> *David Ponzone
> *Sent:* 24. juni 2011 23:42
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] .wav vs .gsm file sizes for 
> recording calls.
>
> Wes,
>
> it's because you are confusing 2 things.
>
> GSM is a codec, so it's a sound format.
>
> WAVE is a kind of container (like MKV) that can contain various codecs.
>
> In your case, you probably did a mistake using sox (it's not an easy 
> tool), so you ended up with a wav which is still GSM audio inside.
>
> You have to tell sox which exact output format you want.
>
> AFAIR, you have to use the -s option (for signed linear).
>
> David Ponzone Direction Technique
>
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>
> tel: 01 74 03 18 97
>
> gsm: 06 66 98 76 34
>
> Service ClientIPeva
>
> tel: 0811 46 26 26
>
> _www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>_- _www.ipeva-studio.com 
> <BLOCKED::http://www.ipeva-studio.com/>_
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
> diffusion non autoris?e est interdite. Tout message ?lectronique est 
> susceptible d'alt?ration. *IPeva* d?cline toute responsabilit? au 
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
> n'?tes pas destinataire de ce message, merci de le d?truire 
> imm?diatement et d'avertir l'exp?diteur./
>
> __
>
>
>
> Le 24/06/2011 ? 23:26, Wes a ?crit :
>
>
>
> In my tests, if I record a call in .wav format, a 10 second file is
> about 177,000 bytes, while a 10 second .gsm file is 17,000 bytes.
>
> I then used sox to convert the .gsm file to a .wav file, and it stayed
> at around 17,000 bytes. So, is the default recording format for .wav
> using a higher sample rate? vs the default conversion format for the sox
> tool?
>
> checking the file type using "file" I see that the larger one is:
> RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
>
> and the wav created by sox via the default conversion from .gsm is:
> RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz
>
> So apparently the larger wav file is 16 bit... how are these recording
> parameters controlled? Can I set it to record directly into the smaller
> wav format? Or will I have to run sox on every file...
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org 
> <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From yehavi.bourvine at gmail.com  Sat Jun 25 09:53:15 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sat, 25 Jun 2011 08:53:15 +0300
Subject: [Freeswitch-users] Recovering from database fault
In-Reply-To: <BANLkTikUftc2ck=AweFLKhksjS6_3fKDUA@mail.gmail.com>
References: <20110625000823.00B8AE80E4@jlo.kiwilink.co.nz>
	<BANLkTikUftc2ck=AweFLKhksjS6_3fKDUA@mail.gmail.com>
Message-ID: <BANLkTik_ZedDsiz7MLP+EPGPeyYrVRk7Dw@mail.gmail.com>

Hello Clive,

  If you've deleted all DB files and FS created fresh ones then all the
registrations information is lost. If you restart the phones (or wait for
the next re-registration) then the phones would register again,

                           __Yehavi:

2011/6/25 curriegrad2004 <curriegrad2004 at gmail.com>

> Are you running FreeSwitch as a non root user? This makes a huge
> difference as I might suspect there are file permission issues that's
> happening here. ie. db folder doesn't allow the fs user account to
> create or write to that directory
>
> On Fri, Jun 24, 2011 at 5:08 PM, Clive Lansink <clive at lansink.co.nz>
> wrote:
> > Yes that's exactly right. You understand the situation correctly.
> >
> > I guess this situation is not going to occur often so I'll just watch to
> see if it ever happens again.
> >
> >
> > Clive Lansink
> > Email: Clive at Lansink.Co.NZ
> > Phone: +64 9 520-4242
> > Mobile: +64 21 663-999
> > Fax: +64 21 789-150
> >
> > -----Original message-----
> > From: Michael Collins <msc at freeswitch.org>
> > To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> > Subject: Re: [Freeswitch-users] Recovering from database fault
> > Reply-to: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> > Date: Fri, 24 Jun 2011 16:36:42 -0700
> >
> > On Fri, Jun 24, 2011 at 4:26 PM, Clive Lansink <clive at lansink.co.nz>
> wrote:
> >
> >> Hello Michael and others.
> >>
> >> I deleted the whold db folder which in our case was ok because we don't
> use
> >> voice mail. Then when I ran Freeswitchconsole, the db folder was
> recreated.
> >> But that was when we had the problem. The phones could make calls but
> none
> >> could ring.
> >>
> >
> > Okay, so you removed the whole db folder then restarted FS and everything
> > worked except for the registrations? Then when you did a complete
> re-install
> > of FS then everything, including registrations, seemed to work?
> >
> > I'd recommend that you update to the latest git and watch and wait. If
> the
> > symptom returns then maybe you could collect a gcore and open a jira,
> unless
> > you're on Windows. (We have Windows guys who can guide you on what to do
> for
> > troubleshooting - I just don't happen to be one of them...)
> >
> > -MC
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From a.luppi at seletech.com  Sat Jun 25 13:10:38 2011
From: a.luppi at seletech.com (Alessandro)
Date: Sat, 25 Jun 2011 11:10:38 +0200
Subject: [Freeswitch-users] Lan redundancy
Message-ID: <4E05A60E.7020207@seletech.com>

Hi,

I'm going to install freeswitch in a system with LAN redundancy 
(duplicated). All the pc have double LAN interfaces. How can I configure 
Freeswitch to work with this configuration? Actually in vars, the 
variable Domains has this value

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


I have to set one ip of the two network interface? I need to set the 
sub-net mask? (example domain=192.168.2.101/255.255.255.0)

Second question:

All the PC with softphone will be connect at two LAN and the two LAN are 
on different Network. (Example one LAN is on network 192.168.2.0 and the 
other in the LAN 192.168.1.0).
I bind the address of one of the two network to freeswitch.
I will add the extension in the internal profile. How does freeswitch 
understand that an extension is in the local network? All the softphone 
should stay on the same network, right?
What happens if an extension configured in the internal profile, try to 
contact FS from a different network?

Best Regards

Alessandro Luppi

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

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From steveayre at gmail.com  Sat Jun 25 14:07:55 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 25 Jun 2011 11:07:55 +0100
Subject: [Freeswitch-users] Lan redundancy
In-Reply-To: <4E05A60E.7020207@seletech.com>
References: <4E05A60E.7020207@seletech.com>
Message-ID: <BANLkTi=+w0fXPd8OZgJik-vPg2JZ5bN=JSR+1NDV7=v7mWQ+Mg@mail.gmail.com>

The way I do this is to use ethernet bonding in active-backup mode:

http://www.kernel.org/doc/Documentation/networking/bonding.txt
http://wiki.debian.org/Bonding

You get a virtual network interface named bond0 which is assigned your IP.
This effectively replaces your eth0. That simplifies the configuration of
everything like FreeSWITCH since you then get a single IP to listen on. The
bonding driver monitors the slave devices (eth0,eth1,etc) and uses an active
one. If a device goes down it automatically switches over to the other.

You can also have different profiles on different IPs, with a profile for
each device, but if a device fails any calls going to that IP will fail
because the signalling/media is still trying to go to that address. Bonding
avoids that problem.

Bonding will probably be enough for you, but for some extra information my
setup is a little more complex than that... that there's redundancy on the
network too - 2 network switches each with 100MBit internet feeds from the
data centre, and interconnected with a 2Gbit trunk, running RSTP. Each
server has one device going to one switch and the other going to the 2nd. It
means that if any switch, device, or cable fails the whole thing will find
another route (even between switches via the data centre's switch if
necessary). A stacked switch would be better, but isn't currently within
budget.

-Steve





On 25 June 2011 10:10, Alessandro <a.luppi at seletech.com> wrote:

> **
> Hi,
>
> I'm going to install freeswitch in a system with LAN redundancy
> (duplicated). All the pc have double LAN interfaces. How can I configure
> Freeswitch to work with this configuration? Actually in vars, the variable
> Domains has this value
>
> <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
>
>
> I have to set one ip of the two network interface? I need to set the
> sub-net mask? (example domain=192.168.2.101/255.255.255.0)
>
> Second question:
>
> All the PC with softphone will be connect at two LAN and the two LAN are on
> different Network. (Example one LAN is on network 192.168.2.0 and the other
> in the LAN 192.168.1.0).
> I bind the address of one of the two network to freeswitch.
> I will add the extension in the internal profile. How does freeswitch
> understand that an extension is in the local network? All the softphone
> should stay on the same network, right?
> What happens if an extension configured in the internal profile, try to
> contact FS from a different network?
>
> Best Regards
>
> Alessandro Luppi
>
> --
> Ing. Alessandro Luppi
> Software development
> Seletech srl
> Via Collodi 8, 20052 Monza (MI) - Italy
> Tel: +39.039.5962000 - Fax: +39.039.9716905
> email: a.luppi at seletech.com - Web: www.seletech.com   or   www.seletech.eu
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jan.berger at video24.no  Sat Jun 25 14:23:52 2011
From: jan.berger at video24.no (Jan Berger)
Date: Sat, 25 Jun 2011 12:23:52 +0200
Subject: [Freeswitch-users] SpiderMonkey 1.8.5
Message-ID: <5D59B0C07A28436F897691CE12F9CAE9@dell9400>

Does anyone know about an existing Visual Studio build for SpiderMonkey
1.8.5?

 

I am interested in testing it out, but I was hoping to find a pre-build or
project for VS 2005, 2008 or 2010.

 

It's only a few months old (march 2011), so might be a bit early yet???

 

Jan

 

 

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From a.luppi at seletech.com  Sat Jun 25 14:55:50 2011
From: a.luppi at seletech.com (Alessandro)
Date: Sat, 25 Jun 2011 12:55:50 +0200
Subject: [Freeswitch-users] Lan redundancy
In-Reply-To: <BANLkTi=+w0fXPd8OZgJik-vPg2JZ5bN=JSR+1NDV7=v7mWQ+Mg@mail.gmail.com>
References: <4E05A60E.7020207@seletech.com>
	<BANLkTi=+w0fXPd8OZgJik-vPg2JZ5bN=JSR+1NDV7=v7mWQ+Mg@mail.gmail.com>
Message-ID: <4E05BEB6.9060306@seletech.com>

Very interesting,

I was thinking to something similar.. I thought to create e third 
virtual interface... So I'll have to use network bonding to all the pc 
in the network...
The two networks will have switch/router (i think hircshmann) so the two 
networks will be the specular image of the other. In the interface named 
bond0 will be configured on the same network of the two network 
interface or I have to create an virtual third network? On the softphone 
for the server proxy I'll set the address of the interface bond0 of the 
server freeswitch right?

Thanks

Alessandro Luppi



Il 25/06/2011 12:07, Steven Ayre ha scritto:
> The way I do this is to use ethernet bonding in active-backup mode:
>
> http://www.kernel.org/doc/Documentation/networking/bonding.txt
> http://wiki.debian.org/Bonding
>
> You get a virtual network interface named bond0 which is assigned your 
> IP. This effectively replaces your eth0. That simplifies the 
> configuration of everything like FreeSWITCH since you then get a 
> single IP to listen on. The bonding driver monitors the slave devices 
> (eth0,eth1,etc) and uses an active one. If a device goes down it 
> automatically switches over to the other.
>
> You can also have different profiles on different IPs, with a profile 
> for each device, but if a device fails any calls going to that IP will 
> fail because the signalling/media is still trying to go to that 
> address. Bonding avoids that problem.
>
> Bonding will probably be enough for you, but for some extra 
> information my setup is a little more complex than that... that 
> there's redundancy on the network too - 2 network switches each with 
> 100MBit internet feeds from the data centre, and interconnected with a 
> 2Gbit trunk, running RSTP. Each server has one device going to one 
> switch and the other going to the 2nd. It means that if any switch, 
> device, or cable fails the whole thing will find another route (even 
> between switches via the data centre's switch if necessary). A stacked 
> switch would be better, but isn't currently within budget.
>
> -Steve
>
>
>
>
>
> On 25 June 2011 10:10, Alessandro <a.luppi at seletech.com 
> <mailto:a.luppi at seletech.com>> wrote:
>
>     Hi,
>
>     I'm going to install freeswitch in a system with LAN redundancy
>     (duplicated). All the pc have double LAN interfaces. How can I
>     configure Freeswitch to work with this configuration? Actually in
>     vars, the variable Domains has this value
>
>     <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
>
>
>     I have to set one ip of the two network interface? I need to set
>     the sub-net mask? (example domain=192.168.2.101/255.255.255.0
>     <http://192.168.2.101/255.255.255.0>)
>
>     Second question:
>
>     All the PC with softphone will be connect at two LAN and the two
>     LAN are on different Network. (Example one LAN is on network
>     192.168.2.0 and the other in the LAN 192.168.1.0).
>     I bind the address of one of the two network to freeswitch.
>     I will add the extension in the internal profile. How does
>     freeswitch understand that an extension is in the local network?
>     All the softphone should stay on the same network, right?
>     What happens if an extension configured in the internal profile,
>     try to contact FS from a different network?
>
>     Best Regards
>
>     Alessandro Luppi
>
>     -- 
>     Ing. Alessandro Luppi
>     Software development
>     Seletech srl
>     Via Collodi 8, 20052 Monza (MI) - Italy
>     Tel: +39.039.5962000 - Fax: +39.039.9716905
>     email:a.luppi at seletech.com  <mailto:a.luppi at seletech.com>  - Web:www.seletech.com  <http://www.seletech.com>    orwww.seletech.eu  <http://www.seletech.eu>
>
>
>     _______________________________________________
>     Join us at ClueCon 2011, Aug 9-11, Chicago
>     http://www.cluecon.com 877-7-4ACLUE
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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

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From steveayre at gmail.com  Sat Jun 25 15:35:46 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 25 Jun 2011 12:35:46 +0100
Subject: [Freeswitch-users] Lan redundancy
In-Reply-To: <4E05BEB6.9060306@seletech.com>
References: <4E05A60E.7020207@seletech.com>
	<BANLkTi=+w0fXPd8OZgJik-vPg2JZ5bN=JSR+1NDV7=v7mWQ+Mg@mail.gmail.com>
	<4E05BEB6.9060306@seletech.com>
Message-ID: <BANLkTinEUrwdVZ2C-8SzmH4ieusrBraFTQ@mail.gmail.com>

bond0 becomes your single network interface. eth0/eth1 aren't used any
longer - they're still there but they're just slaves of bond0.

That means you get a single IP on bond0 (eth0/eth1 don't have an IP) which
is the one FS/softphones would use.

If you're planning to have two independant networks with eth0 on one and
eth1 on the other that'll cause you problems... it's meant to provide the
server with two routes into the same network.

Between devices on the LAN it'll be problematic - in active-backup one
device will be active and the others disabled. There's no guarantee that all
servers'll pick the same slave device though, so some might be on eth0 and
others on eth1 which'd mean they wouldn't be able to reach each other. My 2
switches are connected together and use RSTP to avoid routing loops to avoid
that, which is why I mentioned it in the 1st email.

For the routing to the WAN it'll mean different routers so different IPs,
which'll drop calls with NAT on switchover, and force you to listen to both
public IPs on bond0 / FS if you're not using NAT which'll also mean dropped
calls on switching over.

Our network setup is here: http://pastebin.freeswitch.org/16587, how you
connect to the WAN really will depend on your ISP/Data Centre though.

-Steve



On 25 June 2011 11:55, Alessandro <a.luppi at seletech.com> wrote:

> **
> Very interesting,
>
> I was thinking to something similar.. I thought to create e third virtual
> interface... So I'll have to use network bonding to all the pc in the
> network...
> The two networks will have switch/router (i think hircshmann) so the two
> networks will be the specular image of the other. In the interface named
> bond0 will be configured on the same network of the two network interface or
> I have to create an virtual third network? On the softphone for the server
> proxy I'll set the address of the interface bond0 of the server freeswitch
> right?
>
> Thanks
>
> Alessandro Luppi
>
>

> Il 25/06/2011 12:07, Steven Ayre ha scritto:
>
> The way I do this is to use ethernet bonding in active-backup mode:
>
> http://www.kernel.org/doc/Documentation/networking/bonding.txt
> http://wiki.debian.org/Bonding
>
> You get a virtual network interface named bond0 which is assigned your IP.
> This effectively replaces your eth0. That simplifies the configuration of
> everything like FreeSWITCH since you then get a single IP to listen on. The
> bonding driver monitors the slave devices (eth0,eth1,etc) and uses an active
> one. If a device goes down it automatically switches over to the other.
>
> You can also have different profiles on different IPs, with a profile for
> each device, but if a device fails any calls going to that IP will fail
> because the signalling/media is still trying to go to that address. Bonding
> avoids that problem.
>
> Bonding will probably be enough for you, but for some extra information my
> setup is a little more complex than that... that there's redundancy on the
> network too - 2 network switches each with 100MBit internet feeds from the
> data centre, and interconnected with a 2Gbit trunk, running RSTP. Each
> server has one device going to one switch and the other going to the 2nd. It
> means that if any switch, device, or cable fails the whole thing will find
> another route (even between switches via the data centre's switch if
> necessary). A stacked switch would be better, but isn't currently within
> budget.
>
> -Steve
>
>
>
>
>
> On 25 June 2011 10:10, Alessandro <a.luppi at seletech.com> wrote:
>
>>  Hi,
>>
>> I'm going to install freeswitch in a system with LAN redundancy
>> (duplicated). All the pc have double LAN interfaces. How can I configure
>> Freeswitch to work with this configuration? Actually in vars, the variable
>> Domains has this value
>>
>> <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
>>
>>
>> I have to set one ip of the two network interface? I need to set the
>> sub-net mask? (example domain=192.168.2.101/255.255.255.0)
>>
>> Second question:
>>
>> All the PC with softphone will be connect at two LAN and the two LAN are
>> on different Network. (Example one LAN is on network 192.168.2.0 and the
>> other in the LAN 192.168.1.0).
>> I bind the address of one of the two network to freeswitch.
>> I will add the extension in the internal profile. How does freeswitch
>> understand that an extension is in the local network? All the softphone
>> should stay on the same network, right?
>> What happens if an extension configured in the internal profile, try to
>> contact FS from a different network?
>>
>> Best Regards
>>
>> Alessandro Luppi
>>
>> --
>> Ing. Alessandro Luppi
>> Software development
>> Seletech srl
>> Via Collodi 8, 20052 Monza (MI) - Italy
>> Tel: +39.039.5962000 - Fax: +39.039.9716905
>> email: a.luppi at seletech.com - Web: www.seletech.com   or   www.seletech.eu
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> Ing. Alessandro Luppi
> Software development
> Seletech srl
> Via Collodi 8, 20052 Monza (MI) - Italy
> Tel: +39.039.5962000 - Fax: +39.039.9716905
> email: a.luppi at seletech.com - Web: www.seletech.com   or   www.seletech.eu
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From a.luppi at seletech.com  Sat Jun 25 16:42:37 2011
From: a.luppi at seletech.com (Alessandro)
Date: Sat, 25 Jun 2011 14:42:37 +0200
Subject: [Freeswitch-users] Lan redundancy
In-Reply-To: <BANLkTinEUrwdVZ2C-8SzmH4ieusrBraFTQ@mail.gmail.com>
References: <4E05A60E.7020207@seletech.com>	<BANLkTi=+w0fXPd8OZgJik-vPg2JZ5bN=JSR+1NDV7=v7mWQ+Mg@mail.gmail.com>	<4E05BEB6.9060306@seletech.com>
	<BANLkTinEUrwdVZ2C-8SzmH4ieusrBraFTQ@mail.gmail.com>
Message-ID: <4E05D7BD.3040406@seletech.com>

Ok,

I'll going to talk with who is planning the network to understand 
exactly the architecture of the network. I understand how to manage two 
interfaces in a LAN. I think that with router i'm going to work in a 
WAN.  I think also that the two network are connected so the server is 
reachable from both side and the routing protocol should avoid loops in 
the network.
This is what said to me about network configuration:

http://pastebin.freeswitch.org/16588

I've also another question, I'd like also redundancy on FS server (the 
server machine will be duplicated). I'm going to install 2 server, so if 
one fails the other is ready to go in service. Have you ever try 
something similar?
I was thinking to an heartbeat daemon to manage this situation...

Best Regards

Alessandro Luppi


Il 25/06/2011 13:35, Steven Ayre ha scritto:
> bond0 becomes your single network interface. eth0/eth1 aren't used any 
> longer - they're still there but they're just slaves of bond0.
>
> That means you get a single IP on bond0 (eth0/eth1 don't have an IP) 
> which is the one FS/softphones would use.
>
> If you're planning to have two independant networks with eth0 on one 
> and eth1 on the other that'll cause you problems... it's meant to 
> provide the server with two routes into the same network.
>
> Between devices on the LAN it'll be problematic - in active-backup one 
> device will be active and the others disabled. There's no guarantee 
> that all servers'll pick the same slave device though, so some might 
> be on eth0 and others on eth1 which'd mean they wouldn't be able to 
> reach each other. My 2 switches are connected together and use RSTP to 
> avoid routing loops to avoid that, which is why I mentioned it in the 
> 1st email.
>
> For the routing to the WAN it'll mean different routers so different 
> IPs, which'll drop calls with NAT on switchover, and force you to 
> listen to both public IPs on bond0 / FS if you're not using NAT 
> which'll also mean dropped calls on switching over.
>
> Our network setup is here: http://pastebin.freeswitch.org/16587, how 
> you connect to the WAN really will depend on your ISP/Data Centre though.
>
> -Steve
>
>
>
> On 25 June 2011 11:55, Alessandro <a.luppi at seletech.com 
> <mailto:a.luppi at seletech.com>> wrote:
>
>     Very interesting,
>
>     I was thinking to something similar.. I thought to create e third
>     virtual interface... So I'll have to use network bonding to all
>     the pc in the network...
>     The two networks will have switch/router (i think hircshmann) so
>     the two networks will be the specular image of the other. In the
>     interface named bond0 will be configured on the same network of
>     the two network interface or I have to create an virtual third
>     network? On the softphone for the server proxy I'll set the
>     address of the interface bond0 of the server freeswitch right?
>
>     Thanks
>
>     Alessandro Luppi
>
>
>
>     Il 25/06/2011 12:07, Steven Ayre ha scritto:
>>     The way I do this is to use ethernet bonding in active-backup mode:
>>
>>     http://www.kernel.org/doc/Documentation/networking/bonding.txt
>>     http://wiki.debian.org/Bonding
>>
>>     You get a virtual network interface named bond0 which is assigned
>>     your IP. This effectively replaces your eth0. That simplifies the
>>     configuration of everything like FreeSWITCH since you then get a
>>     single IP to listen on. The bonding driver monitors the slave
>>     devices (eth0,eth1,etc) and uses an active one. If a device goes
>>     down it automatically switches over to the other.
>>
>>     You can also have different profiles on different IPs, with a
>>     profile for each device, but if a device fails any calls going to
>>     that IP will fail because the signalling/media is still trying to
>>     go to that address. Bonding avoids that problem.
>>
>>     Bonding will probably be enough for you, but for some extra
>>     information my setup is a little more complex than that... that
>>     there's redundancy on the network too - 2 network switches each
>>     with 100MBit internet feeds from the data centre, and
>>     interconnected with a 2Gbit trunk, running RSTP. Each server has
>>     one device going to one switch and the other going to the 2nd. It
>>     means that if any switch, device, or cable fails the whole thing
>>     will find another route (even between switches via the data
>>     centre's switch if necessary). A stacked switch would be better,
>>     but isn't currently within budget.
>>
>>     -Steve
>>
>>
>>
>>
>>
>>     On 25 June 2011 10:10, Alessandro <a.luppi at seletech.com
>>     <mailto:a.luppi at seletech.com>> wrote:
>>
>>         Hi,
>>
>>         I'm going to install freeswitch in a system with LAN
>>         redundancy (duplicated). All the pc have double LAN
>>         interfaces. How can I configure Freeswitch to work with this
>>         configuration? Actually in vars, the variable Domains has
>>         this value
>>
>>         <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
>>
>>
>>         I have to set one ip of the two network interface? I need to
>>         set the sub-net mask? (example
>>         domain=192.168.2.101/255.255.255.0
>>         <http://192.168.2.101/255.255.255.0>)
>>
>>         Second question:
>>
>>         All the PC with softphone will be connect at two LAN and the
>>         two LAN are on different Network. (Example one LAN is on
>>         network 192.168.2.0 and the other in the LAN 192.168.1.0).
>>         I bind the address of one of the two network to freeswitch.
>>         I will add the extension in the internal profile. How does
>>         freeswitch understand that an extension is in the local
>>         network? All the softphone should stay on the same network,
>>         right?
>>         What happens if an extension configured in the internal
>>         profile, try to contact FS from a different network?
>>
>>         Best Regards
>>
>>         Alessandro Luppi
>>
>>         -- 
>>         Ing. Alessandro Luppi
>>         Software development
>>         Seletech srl
>>         Via Collodi 8, 20052 Monza (MI) - Italy
>>         Tel: +39.039.5962000 - Fax: +39.039.9716905
>>         email:a.luppi at seletech.com  <mailto:a.luppi at seletech.com>  - Web:www.seletech.com  <http://www.seletech.com>    orwww.seletech.eu  <http://www.seletech.eu>
>>
>>
>>         _______________________________________________
>>         Join us at ClueCon 2011, Aug 9-11, Chicago
>>         http://www.cluecon.com 877-7-4ACLUE
>>
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>     _______________________________________________
>>     Join us at ClueCon 2011, Aug 9-11, Chicago
>>     http://www.cluecon.com  877-7-4ACLUE
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>     -- 
>     Ing. Alessandro Luppi
>     Software development
>     Seletech srl
>     Via Collodi 8, 20052 Monza (MI) - Italy
>     Tel: +39.039.5962000 - Fax: +39.039.9716905
>     email:a.luppi at seletech.com  <mailto:a.luppi at seletech.com>  - Web:www.seletech.com  <http://www.seletech.com>    orwww.seletech.eu  <http://www.seletech.eu>
>
>
>     _______________________________________________
>     Join us at ClueCon 2011, Aug 9-11, Chicago
>     http://www.cluecon.com 877-7-4ACLUE
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com  877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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

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From infos at madovsky.org  Sat Jun 25 19:05:37 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 25 Jun 2011 11:05:37 -0400
Subject: [Freeswitch-users] Lan redundancy
References: <4E05A60E.7020207@seletech.com><BANLkTi=+w0fXPd8OZgJik-vPg2JZ5bN=JSR+1NDV7=v7mWQ+Mg@mail.gmail.com>
	<4E05BEB6.9060306@seletech.com>
Message-ID: <61DFA2F664084D9388ACB5670F9EDE7A@e1705>

I used bond0 with 6 interfaces since one year without problems
  ----- Original Message ----- 
  From: Alessandro 
  To: FreeSWITCH Users Help 
  Sent: Saturday, June 25, 2011 6:55 AM
  Subject: Re: [Freeswitch-users] Lan redundancy


  Very interesting,

  I was thinking to something similar.. I thought to create e third virtual interface... So I'll have to use network bonding to all the pc in the network...
  The two networks will have switch/router (i think hircshmann) so the two networks will be the specular image of the other. In the interface named bond0 will be configured on the same network of the two network interface or I have to create an virtual third network? On the softphone for the server proxy I'll set the address of the interface bond0 of the server freeswitch right?

  Thanks

  Alessandro Luppi  



  Il 25/06/2011 12:07, Steven Ayre ha scritto: 
    The way I do this is to use ethernet bonding in active-backup mode:

    http://www.kernel.org/doc/Documentation/networking/bonding.txt
    http://wiki.debian.org/Bonding

    You get a virtual network interface named bond0 which is assigned your IP. This effectively replaces your eth0. That simplifies the configuration of everything like FreeSWITCH since you then get a single IP to listen on. The bonding driver monitors the slave devices (eth0,eth1,etc) and uses an active one. If a device goes down it automatically switches over to the other.

    You can also have different profiles on different IPs, with a profile for each device, but if a device fails any calls going to that IP will fail because the signalling/media is still trying to go to that address. Bonding avoids that problem.

    Bonding will probably be enough for you, but for some extra information my setup is a little more complex than that... that there's redundancy on the network too - 2 network switches each with 100MBit internet feeds from the data centre, and interconnected with a 2Gbit trunk, running RSTP. Each server has one device going to one switch and the other going to the 2nd. It means that if any switch, device, or cable fails the whole thing will find another route (even between switches via the data centre's switch if necessary). A stacked switch would be better, but isn't currently within budget.

    -Steve






    On 25 June 2011 10:10, Alessandro <a.luppi at seletech.com> wrote:

      Hi,

      I'm going to install freeswitch in a system with LAN redundancy (duplicated). All the pc have double LAN interfaces. How can I configure Freeswitch to work with this configuration? Actually in vars, the variable Domains has this value 
<X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
      I have to set one ip of the two network interface? I need to set the sub-net mask? (example domain=192.168.2.101/255.255.255.0)

      Second question:

      All the PC with softphone will be connect at two LAN and the two LAN are on different Network. (Example one LAN is on network 192.168.2.0 and the other in the LAN 192.168.1.0).
      I bind the address of one of the two network to freeswitch.
      I will add the extension in the internal profile. How does freeswitch understand that an extension is in the local network? All the softphone should stay on the same network, right?
      What happens if an extension configured in the internal profile, try to contact FS from a different network?

      Best Regards

      Alessandro Luppi   


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

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

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




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

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



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


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


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

  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From jan.berger at video24.no  Sun Jun 26 03:30:33 2011
From: jan.berger at video24.no (Jan Berger)
Date: Sun, 26 Jun 2011 01:30:33 +0200
Subject: [Freeswitch-users] SpiderMonkey 1.8.5
In-Reply-To: <5D59B0C07A28436F897691CE12F9CAE9@dell9400>
References: <5D59B0C07A28436F897691CE12F9CAE9@dell9400>
Message-ID: <7E619FFCA9154237AEB776AC6DB3B253@dell9400>

Actually realized that MozillaBuild will build for Visual Studio 2010 and
various other targets directly.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan
Berger
Sent: 25. juni 2011 12:24
To: 'FreeSWITCH Users Help'
Subject: [Freeswitch-users] SpiderMonkey 1.8.5

 

Does anyone know about an existing Visual Studio build for SpiderMonkey
1.8.5?

 

I am interested in testing it out, but I was hoping to find a pre-build or
project for VS 2005, 2008 or 2010.

 

It's only a few months old (march 2011), so might be a bit early yet???

 

Jan

 

 

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From acichocki at supermedia.pl  Sun Jun 26 12:07:10 2011
From: acichocki at supermedia.pl (Artur Cichocki)
Date: Sun, 26 Jun 2011 10:07:10 +0200
Subject: [Freeswitch-users] Removing SDP from 183/180
Message-ID: <4E06E8AE.2080307@supermedia.pl>

Hi All.

I didn't get any response from fs-dev moderator for one month so it goes 
here.

I just wrote this patch. It ads 'sip_remove_183sdp' flag, which removes 
SDP from 183 and returns 180 (it also removes SDP from 180, because FS
changes 180 with SDP to 183).

To turn it on add {sip_remove_183sdp=true} to the dial string.

Why? Because in some situations we don't want to transfer voice between 
clients before a real call establishment (and yes, I know that early 
media in this stage is allowed by RFC).

Real case:
Try to setup two clients, one of them using Sipdroid. Turn on bypass 
mode. Try to call to Sipdroid client, you will hear caller voice 
together with the ring (because Sipdroid sends 180 with SDP, and puts 
voice on the speaker).

Patch attached.

-- 
Artur Cichocki

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From diego.viola at gmail.com  Fri Jun 24 23:36:41 2011
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 24 Jun 2011 15:36:41 -0400
Subject: [Freeswitch-users] Please let's protect the wiki from spammers
Message-ID: <BANLkTinVTUtGPVXCa_3=Lhrpe2esqB2dXw@mail.gmail.com>

Hi,

As most of us may alright know, the FreeSWITCH wiki has been abused
and attacked by spammers, as seen here:
http://wiki.freeswitch.org/wiki/Special:RecentChanges

Some FreeSWITCH users have been deleting and blocking these spammers,
but they are increasing their spamming attacks.

I went into #mediawiki in freenode and asked them if there is any
solution to this and this is what they suggest:

15:18 < bawolff> !spam | diegoviola
15:18 < mw-bot> diegoviola: For information about combating and
handling spam in MediaWiki, see
<http://www.mediawiki.org/wiki/Manual:Combating_spam> and
<http://www.mediawiki.org/wiki/Anti-spam_features>.


I've seen most wikis who have implemented this solution and they no
longer get any spamming after applying said solution. If somebody here
have administration access to http://wiki.freeswitch.org/ I offer my
help to apply this solution.

Thanks,
Diego Viola


From avi at avimarcus.net  Sun Jun 26 18:52:42 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 26 Jun 2011 17:52:42 +0300
Subject: [Freeswitch-users] Please let's protect the wiki from spammers
In-Reply-To: <BANLkTinVTUtGPVXCa_3=Lhrpe2esqB2dXw@mail.gmail.com>
References: <BANLkTinVTUtGPVXCa_3=Lhrpe2esqB2dXw@mail.gmail.com>
Message-ID: <BANLkTi=cK4CUE6jiYAGs3Rb6iQWzRVRtMQ@mail.gmail.com>

Linode uses .htaccess on the edit pages (only those), much like our
pastebin, and is probably the easiest thing to add.
But yes, something needs to be done.. I got sick of deleting spam months
ago.

A captcha on normal accounts to make edits is probably OK, too, as long as
they can manually be moved to "human-trusted".
-Avi



On Fri, Jun 24, 2011 at 10:36 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Hi,
>
> As most of us may alright know, the FreeSWITCH wiki has been abused
> and attacked by spammers, as seen here:
> http://wiki.freeswitch.org/wiki/Special:RecentChanges
>
> Some FreeSWITCH users have been deleting and blocking these spammers,
> but they are increasing their spamming attacks.
>
> I went into #mediawiki in freenode and asked them if there is any
> solution to this and this is what they suggest:
>
> 15:18 < bawolff> !spam | diegoviola
> 15:18 < mw-bot> diegoviola: For information about combating and
> handling spam in MediaWiki, see
> <http://www.mediawiki.org/wiki/Manual:Combating_spam> and
> <http://www.mediawiki.org/wiki/Anti-spam_features>.
>
>
> I've seen most wikis who have implemented this solution and they no
> longer get any spamming after applying said solution. If somebody here
> have administration access to http://wiki.freeswitch.org/ I offer my
> help to apply this solution.
>
> Thanks,
> Diego Viola
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kris at kriskinc.com  Sun Jun 26 21:45:02 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Sun, 26 Jun 2011 13:45:02 -0400
Subject: [Freeswitch-users] Removing SDP from 183/180
In-Reply-To: <4E06E8AE.2080307@supermedia.pl>
References: <4E06E8AE.2080307@supermedia.pl>
Message-ID: <BANLkTimPGGFt9qyF8VGdPnjOc_kKL3TGBA@mail.gmail.com>

Interesting patch for a strange case...

BTW, the "broken" client is the one that generates local ringback when
there is a 180 w/ SDP.

Either way this belongs on Jira:

http://jira.freeswitch.org/

On Sun, Jun 26, 2011 at 4:07 AM, Artur Cichocki <acichocki at supermedia.pl> wrote:
> Hi All.
>
> I didn't get any response from fs-dev moderator for one month so it goes
> here.
>
> I just wrote this patch. It ads 'sip_remove_183sdp' flag, which removes SDP
> from 183 and returns 180 (it also removes SDP from 180, because FS
> changes 180 with SDP to 183).
>
> To turn it on add {sip_remove_183sdp=true} to the dial string.
>
> Why? Because in some situations we don't want to transfer voice between
> clients before a real call establishment (and yes, I know that early media
> in this stage is allowed by RFC).
>
> Real case:
> Try to setup two clients, one of them using Sipdroid. Turn on bypass mode.
> Try to call to Sipdroid client, you will hear caller voice together with the
> ring (because Sipdroid sends 180 with SDP, and puts voice on the speaker).
>
> Patch attached.
>
> --
> Artur Cichocki
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner


From bryansmart at bryansmart.com  Mon Jun 27 00:31:46 2011
From: bryansmart at bryansmart.com (Bryan Smart)
Date: Sun, 26 Jun 2011 16:31:46 -0400
Subject: [Freeswitch-users] Proper prompt gain/level
Message-ID: <FE081E27-8D62-4715-B7AA-64432598E013@bryansmart.com>

As part of creating prompts for my IVRs, I've tried to match the audio gain of my prompts with the gain of the stock English prompts. During this process, I noticed that the English Callie prompts are recorded extremely low (max gain around -16DB). I have the Cepstral Callie voice, and I must set the Cepstral volume to about 50% in order to match the gain of the English prompts. In conferences and other situations where prompts are played over conversations, the level of the prompts are obviously low.

I can, of course, renormalize the gain of the prompts up to -10DB or more with a sound editor. However, I wonder if there is a better way to change the level of the prompts, or if there is a good reason for the prompts to be encoded at such a low level.

I haven't considered all of the implications yet, but I'm fairly sure that encoding the prompts this quietly is not the best approach, even if it is desirable for the prompts to play quietly on a call. For each 6DB reduction in gain, there is a 50% reduction in perceived volume, and one less significant bit is used for storing the audio. In a 16-bit file, a maximum gain of -16 means that only the 14 least significant bits are actually used for encoding the audio. This results in a reduction in dynamic range, but the difference isn't really noticeable as long as the data remains 16-bit. The problem comes when the audio is converted to a different bit depth. For example, most quickie routines for converting 16-bit audio to 8-bit audio will simply chop off the 8 least significant bits. Therefore, when the prompts are converted to 8-bit audio for use by most of the narrow band codecs, the prompts are only using 6 bits of audio. If the volume of the channel is increased, then the 6 bits are promoted, and the dithering errors at the bottom become louder. In the worst case, since these prompts are only encoded with 14 bits of actual data, and converting to an 8-bit channel will only leave 6 bits of actual data, boosting the gain of the prompts on an 8-bit channel to full loudness would result in the noise floor (the level of the dithering crackle) being about -18DB. That's almost as loud as the prompts themselves sound at the moment.

Anyway, regardless of observations, is there a reason why the prompts must be recorded so quietly? If I'd like them louder (without increasing the gain of the entire channel), is there a way other than running them through a tool to renormalize their max levels?

Bryan



From jan.berger at video24.no  Mon Jun 27 01:20:51 2011
From: jan.berger at video24.no (Jan Berger)
Date: Sun, 26 Jun 2011 23:20:51 +0200
Subject: [Freeswitch-users] Proper prompt gain/level
In-Reply-To: <FE081E27-8D62-4715-B7AA-64432598E013@bryansmart.com>
References: <FE081E27-8D62-4715-B7AA-64432598E013@bryansmart.com>
Message-ID: <0D3AEFF37C1643B79CE3F7A7EE9EAEC3@dell9400>

Wavepad is free and will do bulk changes to sound files.

Won't AGC solve this?

Jan

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bryan
Smart
Sent: 26. juni 2011 22:32
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Proper prompt gain/level

As part of creating prompts for my IVRs, I've tried to match the audio gain
of my prompts with the gain of the stock English prompts. During this
process, I noticed that the English Callie prompts are recorded extremely
low (max gain around -16DB). I have the Cepstral Callie voice, and I must
set the Cepstral volume to about 50% in order to match the gain of the
English prompts. In conferences and other situations where prompts are
played over conversations, the level of the prompts are obviously low.

I can, of course, renormalize the gain of the prompts up to -10DB or more
with a sound editor. However, I wonder if there is a better way to change
the level of the prompts, or if there is a good reason for the prompts to be
encoded at such a low level.

I haven't considered all of the implications yet, but I'm fairly sure that
encoding the prompts this quietly is not the best approach, even if it is
desirable for the prompts to play quietly on a call. For each 6DB reduction
in gain, there is a 50% reduction in perceived volume, and one less
significant bit is used for storing the audio. In a 16-bit file, a maximum
gain of -16 means that only the 14 least significant bits are actually used
for encoding the audio. This results in a reduction in dynamic range, but
the difference isn't really noticeable as long as the data remains 16-bit.
The problem comes when the audio is converted to a different bit depth. For
example, most quickie routines for converting 16-bit audio to 8-bit audio
will simply chop off the 8 least significant bits. Therefore, when the
prompts are converted to 8-bit audio for use by most of the narrow band
codecs, the prompts are only using 6 bits of audio. If the volume of the
channel is increased, then the 
 6 bits are promoted, and the dithering errors at the bottom become louder.
In the worst case, since these prompts are only encoded with 14 bits of
actual data, and converting to an 8-bit channel will only leave 6 bits of
actual data, boosting the gain of the prompts on an 8-bit channel to full
loudness would result in the noise floor (the level of the dithering
crackle) being about -18DB. That's almost as loud as the prompts themselves
sound at the moment.

Anyway, regardless of observations, is there a reason why the prompts must
be recorded so quietly? If I'd like them louder (without increasing the gain
of the entire channel), is there a way other than running them through a
tool to renormalize their max levels?

Bryan


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

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



From gcd at i.ph  Mon Jun 27 03:04:24 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Mon, 27 Jun 2011 07:04:24 +0800
Subject: [Freeswitch-users] Proper prompt gain/level
In-Reply-To: <0D3AEFF37C1643B79CE3F7A7EE9EAEC3@dell9400>
References: <FE081E27-8D62-4715-B7AA-64432598E013@bryansmart.com>
	<0D3AEFF37C1643B79CE3F7A7EE9EAEC3@dell9400>
Message-ID: <BANLkTinrhFqqvHz_J1sO-OQvseGnyiZ05Q@mail.gmail.com>

you can also run sox on a script to change the volume of the sound files by
batch.

On Mon, Jun 27, 2011 at 5:20 AM, Jan Berger <jan.berger at video24.no> wrote:

> Wavepad is free and will do bulk changes to sound files.
>
> Won't AGC solve this?
>
> Jan
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bryan
> Smart
> Sent: 26. juni 2011 22:32
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Proper prompt gain/level
>
> As part of creating prompts for my IVRs, I've tried to match the audio gain
> of my prompts with the gain of the stock English prompts. During this
> process, I noticed that the English Callie prompts are recorded extremely
> low (max gain around -16DB). I have the Cepstral Callie voice, and I must
> set the Cepstral volume to about 50% in order to match the gain of the
> English prompts. In conferences and other situations where prompts are
> played over conversations, the level of the prompts are obviously low.
>
> I can, of course, renormalize the gain of the prompts up to -10DB or more
> with a sound editor. However, I wonder if there is a better way to change
> the level of the prompts, or if there is a good reason for the prompts to
> be
> encoded at such a low level.
>
> I haven't considered all of the implications yet, but I'm fairly sure that
> encoding the prompts this quietly is not the best approach, even if it is
> desirable for the prompts to play quietly on a call. For each 6DB reduction
> in gain, there is a 50% reduction in perceived volume, and one less
> significant bit is used for storing the audio. In a 16-bit file, a maximum
> gain of -16 means that only the 14 least significant bits are actually used
> for encoding the audio. This results in a reduction in dynamic range, but
> the difference isn't really noticeable as long as the data remains 16-bit.
> The problem comes when the audio is converted to a different bit depth. For
> example, most quickie routines for converting 16-bit audio to 8-bit audio
> will simply chop off the 8 least significant bits. Therefore, when the
> prompts are converted to 8-bit audio for use by most of the narrow band
> codecs, the prompts are only using 6 bits of audio. If the volume of the
> channel is increased, then the
>  6 bits are promoted, and the dithering errors at the bottom become louder.
> In the worst case, since these prompts are only encoded with 14 bits of
> actual data, and converting to an 8-bit channel will only leave 6 bits of
> actual data, boosting the gain of the prompts on an 8-bit channel to full
> loudness would result in the noise floor (the level of the dithering
> crackle) being about -18DB. That's almost as loud as the prompts themselves
> sound at the moment.
>
> Anyway, regardless of observations, is there a reason why the prompts must
> be recorded so quietly? If I'd like them louder (without increasing the
> gain
> of the entire channel), is there a way other than running them through a
> tool to renormalize their max levels?
>
> Bryan
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kheimerl at cs.berkeley.edu  Mon Jun 27 05:23:06 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Sun, 26 Jun 2011 18:23:06 -0700
Subject: [Freeswitch-users] Reject SIP registrations
Message-ID: <BANLkTikU6fRYwmUTxzsSHyWs8raukLx4jw@mail.gmail.com>

Hello FS Users!

I'm trying to create the following setup. When a user registers, if
they register on a known account (lets say X), they do not need a
password. X's registration is immediately OK'd, and everything is
great. I've gotten that working using the ACL. The IP address of our
SIP clients are added through cidr and the clients do not need to give
passwords.

However, for some reason, if another account that does not exist in
the directory (let's say Y) registers, FS returns with a 200 OK,
instead of rejecting Y. I'm trying to figure out why this is the case,
and how to remedy that fact.

I have the following line in my internal.xml file, which I had assumed
would force this function:
   <!-- Force the user and auth-user to match. -->
   <param name="inbound-reg-force-matching-username" value="true"/>

However, it does not work. In my directory, each individual account as
the following lines:
  <user id="1303">
    <params>
      <param name="accept-blind-reg" value="true"/>
      <param name="accept-blind-auth" value="true"/>
      <param name="vm-password" value="1000"/>
    </params>

Though I've found that removing it (from all users in the directory)
doesn't help.

I'm primarily concerned with the line in internal.xml; it seems
possible that the fact that we do not have an auth-user (because we do
not require auth) means that this won't work. However, I have yet to
test that hypothesis. The ACL has been the most confusing aspect of
this installation, with a lot of undocumented aspects, and I get the
nagging feeling this is another. I could very well be wrong though.

Thanks for any direction.


From sid.kshatriya at gmail.com  Mon Jun 27 09:52:01 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Mon, 27 Jun 2011 11:22:01 +0530
Subject: [Freeswitch-users] Proper prompt gain/level
In-Reply-To: <BANLkTinrhFqqvHz_J1sO-OQvseGnyiZ05Q@mail.gmail.com>
References: <FE081E27-8D62-4715-B7AA-64432598E013@bryansmart.com>
	<0D3AEFF37C1643B79CE3F7A7EE9EAEC3@dell9400>
	<BANLkTinrhFqqvHz_J1sO-OQvseGnyiZ05Q@mail.gmail.com>
Message-ID: <BANLkTim5QrkH9=2107-Hnd8XHu7c3-sV-g@mail.gmail.com>

Yeah sox is great for command line and batch changes to volume levels

// this command invocation outputs the maximum volume increase
<name-of-input-sound-file> can go through without any distortion. The
maximum volume increase possible is stored as a decimal in max_vol.txt
sox <name-of-input-sound-file> -n stat -v 2> max_vol.txt
// This command actually boosts the volume level of the input file. Output
is stored in <name-of-output-sound-file>
sox -v `cat max_vol.txt` <name-of-input-sound-file>
<name-of-output-sound-file>

On Mon, Jun 27, 2011 at 4:34 AM, Nandy Dagondon <gcd at i.ph> wrote:

> you can also run sox on a script to change the volume of the sound files by
> batch.
>
> On Mon, Jun 27, 2011 at 5:20 AM, Jan Berger <jan.berger at video24.no> wrote:
>
>> Wavepad is free and will do bulk changes to sound files.
>>
>> Won't AGC solve this?
>>
>> Jan
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bryan
>> Smart
>> Sent: 26. juni 2011 22:32
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: [Freeswitch-users] Proper prompt gain/level
>>
>> As part of creating prompts for my IVRs, I've tried to match the audio
>> gain
>> of my prompts with the gain of the stock English prompts. During this
>> process, I noticed that the English Callie prompts are recorded extremely
>> low (max gain around -16DB). I have the Cepstral Callie voice, and I must
>> set the Cepstral volume to about 50% in order to match the gain of the
>> English prompts. In conferences and other situations where prompts are
>> played over conversations, the level of the prompts are obviously low.
>>
>> I can, of course, renormalize the gain of the prompts up to -10DB or more
>> with a sound editor. However, I wonder if there is a better way to change
>> the level of the prompts, or if there is a good reason for the prompts to
>> be
>> encoded at such a low level.
>>
>> I haven't considered all of the implications yet, but I'm fairly sure that
>> encoding the prompts this quietly is not the best approach, even if it is
>> desirable for the prompts to play quietly on a call. For each 6DB
>> reduction
>> in gain, there is a 50% reduction in perceived volume, and one less
>> significant bit is used for storing the audio. In a 16-bit file, a maximum
>> gain of -16 means that only the 14 least significant bits are actually
>> used
>> for encoding the audio. This results in a reduction in dynamic range, but
>> the difference isn't really noticeable as long as the data remains 16-bit.
>> The problem comes when the audio is converted to a different bit depth.
>> For
>> example, most quickie routines for converting 16-bit audio to 8-bit audio
>> will simply chop off the 8 least significant bits. Therefore, when the
>> prompts are converted to 8-bit audio for use by most of the narrow band
>> codecs, the prompts are only using 6 bits of audio. If the volume of the
>> channel is increased, then the
>>  6 bits are promoted, and the dithering errors at the bottom become
>> louder.
>> In the worst case, since these prompts are only encoded with 14 bits of
>> actual data, and converting to an 8-bit channel will only leave 6 bits of
>> actual data, boosting the gain of the prompts on an 8-bit channel to full
>> loudness would result in the noise floor (the level of the dithering
>> crackle) being about -18DB. That's almost as loud as the prompts
>> themselves
>> sound at the moment.
>>
>> Anyway, regardless of observations, is there a reason why the prompts must
>> be recorded so quietly? If I'd like them louder (without increasing the
>> gain
>> of the entire channel), is there a way other than running them through a
>> tool to renormalize their max levels?
>>
>> Bryan
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sidharth Kshatriya
www.sidk.info
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From mays.david at gmail.com  Mon Jun 27 10:27:19 2011
From: mays.david at gmail.com (David Ma)
Date: Mon, 27 Jun 2011 14:27:19 +0800
Subject: [Freeswitch-users] Leg-A is automatically disconnected on Leg-B
 orginate failure
In-Reply-To: <BANLkTimxBU8m311LVKMWcFOjukwLg+ZUGQ@mail.gmail.com>
References: <1308213151543-6482192.post@n2.nabble.com>
	<BANLkTi=xYgBTPVGxVGmk9f2=bmX5X2vF=Q@mail.gmail.com>
	<BANLkTikhPmjN5HeeJHBnQt1ivA8zbtCnYA@mail.gmail.com>
	<BANLkTi=m7oMix1ZRC9URtRuAbsG2fWyhMQ@mail.gmail.com>
	<BANLkTimxBU8m311LVKMWcFOjukwLg+ZUGQ@mail.gmail.com>
Message-ID: <BANLkTinZJf3BxB7QHefnmOvxQmVW2r4dTg@mail.gmail.com>

Hi Michael,

Thanks very much for the response. There is no dialplan specified for this
call. In the FS setting, G729 is used as preferred stack for originating
calls. The leg-A and B are bridged immediately after receiving
PROGRESS-MEDIA from leg-B.

Calls are originated with following parameters:
*
-- Leg-A --
            "api originate
{origination_caller_id_number=<caller>,sip_cid_type=pid,privacy=yes,continue_on_fail=true}
<called> &park()"

**-- Leg-B --*
*            "api originate
{origination_caller_id_number=<caller>,originate_timeout=60,sip_cid_type=pid,privacy=yes,continue_on_fail=false}
<called> &park()"
*

The entire debug log for this call follows.

Thanks,
D.Ma

=================================
2011-06-24 13:29:35.887830 [DEBUG] switch_ivr_originate.c:1971 variable
string 0 = [origination_caller_id_number=03996563750911]
2011-06-24 13:29:35.887830 [DEBUG] switch_ivr_originate.c:1971 variable
string 1 = [originate_timeout=60]
2011-06-24 13:29:35.887830 [DEBUG] switch_ivr_originate.c:1971 variable
string 2 = [ccd_session_id=20110624132936888918]
2011-06-24 13:29:35.887830 [DEBUG] switch_ivr_originate.c:1971 variable
string 3 = [sip_cid_type=pid]
2011-06-24 13:29:35.887830 [DEBUG] switch_ivr_originate.c:1971 variable
string 4 = [privacy=yes]
2011-06-24 13:29:35.887830 [DEBUG] switch_ivr_originate.c:1971 variable
string 5 = [continue_on_fail=true]
2011-06-24 13:29:35.887830 [NOTICE] switch_channel.c:808 New Channel
sofia/external/03996597632298 at 203.208.207.212[c0bd700d-913c-42ad-b68f-81001bf658b8]
2011-06-24 13:29:35.887830 [DEBUG] mod_sofia.c:4129 (sofia/external/
03996597632298 at 203.208.207.212) State Change CS_NEW -> CS_INIT
2011-06-24 13:29:35.887830 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996597632298 at 203.208.207.212 [BREAK]
2011-06-24 13:29:35.889072 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996597632298 at 203.208.207.212) Running State Change CS_INIT
2011-06-24 13:29:35.889829 [DEBUG] switch_core_state_machine.c:356
(sofia/external/03996597632298 at 203.208.207.212) State INIT
2011-06-24 13:29:35.889829 [DEBUG] mod_sofia.c:84 sofia/external/
03996597632298 at 203.208.207.212 SOFIA INIT
send 999 bytes to udp/[203.208.207.212]:5060 at 05:29:36.398490:
   ------------------------------------------------------------------------
   INVITE sip:03996597632298 at 203.208.207.212 SIP/2.0
   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKtpaQ28SQXKvem
   Max-Forwards: 70
   From: "" <sip:03996563750911 at 202.73.56.46>;tag=5SNjUy956H2mm
   To: <sip:03996597632298 at 203.208.207.212>
   Call-ID: ca5d29b7-18c5-122f-3387-0015c5fc7ad9
   CSeq: 14110624 INVITE
   Contact: <sip:mod_sofia at 202.73.56.46:5080>
   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, PRACK, NOTIFY
   Supported: 100rel, timer, precondition, path, replaces
   Allow-Events: talk, hold, refer
   Privacy: none
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 204
   X-FS-Support: update_display
   P-Asserted-Identity: "" <sip:03996563750911 at 202.73.56.46>

   v=0
   o=FreeSWITCH 1308862405 1308862406 IN IP4 202.73.56.46
   s=FreeSWITCH
   c=IN IP4 202.73.56.46
   t=0 0
   m=audio 30970 RTP/AVP 18 3 101 13
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
2011-06-24 13:29:35.891343 [DEBUG] mod_sofia.c:124 (sofia/external/
03996597632298 at 203.208.207.212) State Change CS_INIT -> CS_ROUTING
2011-06-24 13:29:35.891343 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996597632298 at 203.208.207.212 [BREAK]
2011-06-24 13:29:35.891343 [DEBUG] sofia.c:4646 Channel sofia/external/
03996597632298 at 203.208.207.212 entering state [calling][0]
2011-06-24 13:29:35.891343 [DEBUG] switch_core_state_machine.c:356
(sofia/external/03996597632298 at 203.208.207.212) State INIT going to sleep
2011-06-24 13:29:35.891343 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996597632298 at 203.208.207.212) Running State Change
CS_ROUTING
2011-06-24 13:29:35.891343 [DEBUG] switch_channel.c:1657 (sofia/external/
03996597632298 at 203.208.207.212) Callstate Change DOWN -> RINGING
2011-06-24 13:29:35.891343 [DEBUG] switch_core_state_machine.c:359
(sofia/external/03996597632298 at 203.208.207.212) State ROUTING
2011-06-24 13:29:35.891343 [DEBUG] mod_sofia.c:147 sofia/external/
03996597632298 at 203.208.207.212 SOFIA ROUTING
2011-06-24 13:29:35.891343 [DEBUG] switch_ivr_originate.c:66
(sofia/external/03996597632298 at 203.208.207.212) State Change CS_ROUTING ->
CS_CONSUME_MEDIA
2011-06-24 13:29:35.891343 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996597632298 at 203.208.207.212 [BREAK]
2011-06-24 13:29:35.891343 [DEBUG] switch_core_state_machine.c:359
(sofia/external/03996597632298 at 203.208.207.212) State ROUTING going to sleep
2011-06-24 13:29:35.891343 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996597632298 at 203.208.207.212) Running State Change
CS_CONSUME_MEDIA
2011-06-24 13:29:35.891343 [DEBUG] switch_core_state_machine.c:378
(sofia/external/03996597632298 at 203.208.207.212) State CONSUME_MEDIA
2011-06-24 13:29:35.891343 [DEBUG] switch_core_state_machine.c:378
(sofia/external/03996597632298 at 203.208.207.212) State CONSUME_MEDIA going to
sleep
recv 307 bytes from udp/[203.208.207.212]:5060 at 05:29:36.404598:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.1.1.46:5080
;rport=5080;branch=z9hG4bKtpaQ28SQXKvem;received=10.1.1.46
   From: "" <sip:03996563750911 at 10.1.1.46>;tag=5SNjUy956H2mm
   To: <sip:03996597632298 at 203.208.207.212>
   Call-ID: ca5d29b7-18c5-122f-3387-0015c5fc7ad9
   CSeq: 14110624 INVITE
   Content-Length:     0

   ------------------------------------------------------------------------
recv 774 bytes from udp/[203.208.207.212]:5060 at 05:29:37.615248:
   ------------------------------------------------------------------------
   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP 10.1.1.46:5080
;rport=5080;branch=z9hG4bKtpaQ28SQXKvem;received=10.1.1.46
   From: "" <sip:03996563750911 at 10.1.1.46>;tag=5SNjUy956H2mm
   To: <sip:03996597632298 at 203.208.207.212
>;tag=2QGB951HCR30000E1D0001Ll000B8RX0S6R1TX
   Call-ID: ca5d29b7-18c5-122f-3387-0015c5fc7ad9
   CSeq: 14110624 INVITE
   Contact: <sip:203.208.207.212:5060>
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
SUBSCRIBE, UPDATE
   Content-Type: application/sdp
   Require: 100rel
   RSeq: 1
   Content-Length:   231

   v=0
   o=- 508539113671071081 1 IN IP4 203.208.207.212
   s=session
   c=IN IP4 203.208.207.201
   t=0 0
   m=audio 25120 RTP/AVP 18 101
   a=rtpmap:18 G729/8000
   a=fmtp:18 annexb=no
   a=rtpmap:101 telephone-event/8000
   a=ptime:20
   a=sendrecv
   ------------------------------------------------------------------------
send 651 bytes to udp/[203.208.207.212]:5060 at 05:29:37.615605:
   ------------------------------------------------------------------------
   PRACK sip:203.208.207.212:5060 SIP/2.0
   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKUZ3F43aUtvj1F
   Max-Forwards: 70
   From: "" <sip:03996563750911 at 202.73.56.46>;tag=5SNjUy956H2mm
   To: <sip:03996597632298 at 203.208.207.212
>;tag=2QGB951HCR30000E1D0001Ll000B8RX0S6R1TX
   Call-ID: ca5d29b7-18c5-122f-3387-0015c5fc7ad9
   CSeq: 14110625 PRACK
   Contact: <sip:mod_sofia at 202.73.56.46:5080>
   RAck: 1 14110624 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, PRACK, NOTIFY
   Supported: 100rel, timer, precondition, path, replaces
   Content-Length: 0

   ------------------------------------------------------------------------
2011-06-24 13:29:37.107576 [INFO] sofia.c:729 sofia/external/
03996597632298 at 203.208.207.212 Update Callee ID to "Outbound Call"
<03996597632298>
2011-06-24 13:29:37.107576 [DEBUG] sofia.c:4646 Channel sofia/external/
03996597632298 at 203.208.207.212 entering state [proceeding][183]
2011-06-24 13:29:37.107576 [DEBUG] sofia.c:4657 Remote SDP:
v=0
o=- 508539113671071081 1 IN IP4 203.208.207.212
s=session
c=IN IP4 203.208.207.201
t=0 0
m=audio 25120 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=ptime:20

2011-06-24 13:29:37.107576 [DEBUG] sofia_glue.c:4467 Audio Codec Compare
[G729:18:8000:20:8000]/[G729:18:8000:20:8000]
2011-06-24 13:29:37.107576 [DEBUG] sofia_glue.c:2757 Set Codec
sofia/external/03996597632298 at 203.208.207.212 G729/8000 20 ms 160 samples
8000 bits
2011-06-24 13:29:37.108995 [DEBUG] sofia_glue.c:4565 Set 2833 dtmf send
payload to 101
2011-06-24 13:29:37.108995 [DEBUG] sofia_glue.c:2987 AUDIO RTP
[sofia/external/03996597632298 at 203.208.207.212] 10.1.1.46 port 30970 ->
203.208.207.201 port 25120 codec: 18 ms: 20
2011-06-24 13:29:37.108995 [DEBUG] switch_rtp.c:1607 Starting timer [soft]
160 bytes per 20ms
2011-06-24 13:29:37.109893 [DEBUG] sofia_glue.c:3228 Set 2833 dtmf send
payload to 101
2011-06-24 13:29:37.109893 [DEBUG] sofia_glue.c:3233 Set 2833 dtmf receive
payload to 101
2011-06-24 13:29:37.109893 [NOTICE] sofia_glue.c:3680 Pre-Answer
sofia/external/03996597632298 at 203.208.207.212!
2011-06-24 13:29:37.109893 [DEBUG] switch_channel.c:2627 (sofia/external/
03996597632298 at 203.208.207.212) Callstate Change RINGING -> EARLY
2011-06-24 13:29:37.109893 [DEBUG] switch_ivr_originate.c:3408 Originate
Resulted in Success: [sofia/external/03996597632298 at 203.208.207.212]
2011-06-24 13:29:37.109893 [DEBUG] mod_commands.c:3205 (sofia/external/
03996597632298 at 203.208.207.212) State Change CS_CONSUME_MEDIA -> CS_EXECUTE
2011-06-24 13:29:37.109893 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996597632298 at 203.208.207.212 [BREAK]
2011-06-24 13:29:37.111618 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996597632298 at 203.208.207.212) Running State Change
CS_EXECUTE
2011-06-24 13:29:37.111618 [DEBUG] switch_core_state_machine.c:366
(sofia/external/03996597632298 at 203.208.207.212) State EXECUTE
2011-06-24 13:29:37.111618 [DEBUG] mod_sofia.c:240 sofia/external/
03996597632298 at 203.208.207.212 SOFIA EXECUTE
2011-06-24 13:29:37.111618 [DEBUG] switch_core_state_machine.c:157
sofia/external/03996597632298 at 203.208.207.212 Standard EXECUTE
EXECUTE sofia/external/03996597632298 at 203.208.207.212 park()
recv 382 bytes from udp/[203.208.207.212]:5060 at 05:29:37.620736:
   ------------------------------------------------------------------------
   SIP/2.0 200 Ok
   Via: SIP/2.0/UDP 10.1.1.46:5080
;rport=5080;branch=z9hG4bKUZ3F43aUtvj1F;received=10.1.1.46
   From: "" <sip:03996563750911 at 10.1.1.46>;tag=5SNjUy956H2mm
   To: <sip:03996597632298 at 203.208.207.212
>;tag=2QGB951HCR30000E1D0001Ll000B8RX0S6R1TX
   Call-ID: ca5d29b7-18c5-122f-3387-0015c5fc7ad9
   CSeq: 14110625 PRACK
   Contact: <sip:203.208.207.212:5060>
   Content-Length:     0

   ------------------------------------------------------------------------
2011-06-24 13:29:37.160714 [DEBUG] switch_rtp.c:2933 Correct ip/port
confirmed.
recv 512 bytes from udp/[203.208.207.212]:5060 at 05:29:45.146388:
   ------------------------------------------------------------------------
   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP 10.1.1.46:5080
;rport=5080;branch=z9hG4bKtpaQ28SQXKvem;received=10.1.1.46
   From: "" <sip:03996563750911 at 10.1.1.46>;tag=5SNjUy956H2mm
   To: <sip:03996597632298 at 203.208.207.212
>;tag=2QGB951HCR30000E1D0001Ll000B8RX0S6R1TX
   Call-ID: ca5d29b7-18c5-122f-3387-0015c5fc7ad9
   CSeq: 14110624 INVITE
   Contact: <sip:203.208.207.212:5060>
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
SUBSCRIBE, UPDATE
   Require: 100rel
   RSeq: 2
   Content-Length:     0

   ------------------------------------------------------------------------
send 651 bytes to udp/[203.208.207.212]:5060 at 05:29:45.146628:
   ------------------------------------------------------------------------
   PRACK sip:203.208.207.212:5060 SIP/2.0
   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKv8v85yUyQ58KB
  Max-Forwards: 70
   From: "" <sip:03996563750911 at 202.73.56.46>;tag=5SNjUy956H2mm
   To: <sip:03996597632298 at 203.208.207.212
>;tag=2QGB951HCR30000E1D0001Ll000B8RX0S6R1TX
   Call-ID: ca5d29b7-18c5-122f-3387-0015c5fc7ad9
   CSeq: 14110626 PRACK
   Contact: <sip:mod_sofia at 202.73.56.46:5080>
   RAck: 2 14110624 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, PRACK, NOTIFY
   Supported: 100rel, timer, precondition, path, replaces
   Content-Length: 0

   ------------------------------------------------------------------------
2011-06-24 13:29:44.639310 [INFO] sofia.c:729 sofia/external/
03996597632298 at 203.208.207.212 Update Callee ID to "03996597632298"
<03996597632298>
2011-06-24 13:29:44.639310 [DEBUG] sofia.c:4641 Channel sofia/external/
03996597632298 at 203.208.207.212 skipping state [proceeding][183]
recv 382 bytes from udp/[203.208.207.212]:5060 at 05:29:45.151718:
   ------------------------------------------------------------------------
   SIP/2.0 200 Ok
   Via: SIP/2.0/UDP 10.1.1.46:5080
;rport=5080;branch=z9hG4bKv8v85yUyQ58KB;received=10.1.1.46
   From: "" <sip:03996563750911 at 10.1.1.46>;tag=5SNjUy956H2mm
   To: <sip:03996597632298 at 203.208.207.212
>;tag=2QGB951HCR30000E1D0001Ll000B8RX0S6R1TX
   Call-ID: ca5d29b7-18c5-122f-3387-0015c5fc7ad9
   CSeq: 14110626 PRACK
   Contact: <sip:203.208.207.212:5060>
   Content-Length:     0

   ------------------------------------------------------------------------
recv 529 bytes from udp/[203.208.207.212]:5060 at 05:29:47.446072:
   ------------------------------------------------------------------------
   SIP/2.0 200 Ok
   Via: SIP/2.0/UDP 10.1.1.46:5080
;rport=5080;branch=z9hG4bKtpaQ28SQXKvem;received=10.1.1.46
   From: "" <sip:03996563750911 at 10.1.1.46>;tag=5SNjUy956H2mm
   To: <sip:03996597632298 at 203.208.207.212
>;tag=2QGB951HCR30000E1D0001Ll000B8RX0S6R1TX
   Call-ID: ca5d29b7-18c5-122f-3387-0015c5fc7ad9
   CSeq: 14110624 INVITE
   Contact: <sip:203.208.207.212:5060>
   Allow-Events: refer
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
SUBSCRIBE, UPDATE
   Supported: 100rel, timer, replaces
   Content-Length:     0

   ------------------------------------------------------------------------
2011-06-24 13:29:46.938477 [DEBUG] sofia.c:4646 Channel sofia/external/
03996597632298 at 203.208.207.212 entering state [completing][200]
send 405 bytes to udp/[203.208.207.212]:5060 at 05:29:47.446869:
   ------------------------------------------------------------------------
   ACK sip:203.208.207.212:5060 SIP/2.0
   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKXHp17Sc2meZ6p
   Max-Forwards: 70
   From: "" <sip:03996563750911 at 202.73.56.46>;tag=5SNjUy956H2mm
   To: <sip:03996597632298 at 203.208.207.212
>;tag=2QGB951HCR30000E1D0001Ll000B8RX0S6R1TX
   Call-ID: ca5d29b7-18c5-122f-3387-0015c5fc7ad9
   CSeq: 14110624 ACK
   Contact: <sip:mod_sofia at 202.73.56.46:5080>
   Content-Length: 0

   ------------------------------------------------------------------------
2011-06-24 13:29:46.939738 [DEBUG] sofia.c:4646 Channel sofia/external/
03996597632298 at 203.208.207.212 entering state [ready][200]
2011-06-24 13:29:46.939738 [DEBUG] switch_channel.c:2782 (sofia/external/
03996597632298 at 203.208.207.212) Callstate Change EARLY -> ACTIVE
2011-06-24 13:29:46.939738 [NOTICE] sofia.c:5175 Channel [sofia/external/
03996597632298 at 203.208.207.212] has been answered
2011-06-24 13:29:46.942763 [DEBUG] switch_scheduler.c:214 Added task 27
switch_ivr_schedule_hangup (c0bd700d-913c-42ad-b68f-81001bf658b8) to run at
1308896986
2011-06-24 13:29:46.944563 [DEBUG] switch_core_session.c:954 Send signal
sofia/external/03996597632298 at 203.208.207.212 [BREAK]
2011-06-24 13:29:46.945633 [DEBUG] switch_ivr_originate.c:1971 variable
string 0 = [origination_caller_id_number=03996597632298]
2011-06-24 13:29:46.945633 [DEBUG] switch_ivr_originate.c:1971 variable
string 1 = [ccd_session_id=20110624132936888918]
2011-06-24 13:29:46.945633 [DEBUG] switch_ivr_originate.c:1971 variable
string 2 = [sip_cid_type=pid]
2011-06-24 13:29:46.945633 [DEBUG] switch_ivr_originate.c:1971 variable
string 3 = [privacy=yes]
2011-06-24 13:29:46.945633 [DEBUG] switch_ivr_originate.c:1971 variable
string 4 = [continue_on_fail=false]
2011-06-24 13:29:46.945633 [NOTICE] switch_channel.c:808 New Channel
sofia/external/03996563750911 at 203.208.207.212[817124e6-26fe-46cc-af55-89715abdfced]
2011-06-24 13:29:46.945633 [DEBUG] mod_sofia.c:4129 (sofia/external/
03996563750911 at 203.208.207.212) State Change CS_NEW -> CS_INIT
2011-06-24 13:29:46.945633 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996563750911 at 203.208.207.212 [BREAK]
2011-06-24 13:29:46.945633 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996563750911 at 203.208.207.212) Running State Change CS_INIT
2011-06-24 13:29:46.945633 [DEBUG] switch_core_state_machine.c:356
(sofia/external/03996563750911 at 203.208.207.212) State INIT
2011-06-24 13:29:46.945633 [DEBUG] mod_sofia.c:84 sofia/external/
03996563750911 at 203.208.207.212 SOFIA INIT
2011-06-24 13:29:46.945633 [DEBUG] mod_sofia.c:124 (sofia/external/
03996563750911 at 203.208.207.212) State Change CS_INIT -> CS_ROUTING
2011-06-24 13:29:46.945633 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996563750911 at 203.208.207.212 [BREAK]
2011-06-24 13:29:46.945633 [DEBUG] switch_core_state_machine.c:356
(sofia/external/03996563750911 at 203.208.207.212) State INIT going to sleep
2011-06-24 13:29:46.945633 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996563750911 at 203.208.207.212) Running State Change
CS_ROUTING
2011-06-24 13:29:46.945633 [DEBUG] switch_channel.c:1657 (sofia/external/
03996563750911 at 203.208.207.212) Callstate Change DOWN -> RINGING
2011-06-24 13:29:46.945633 [DEBUG] switch_core_state_machine.c:359
(sofia/external/03996563750911 at 203.208.207.212) State ROUTING
2011-06-24 13:29:46.945633 [DEBUG] mod_sofia.c:147 sofia/external/
03996563750911 at 203.208.207.212 SOFIA ROUTING
2011-06-24 13:29:46.945633 [DEBUG] switch_ivr_originate.c:66
(sofia/external/03996563750911 at 203.208.207.212) State Change CS_ROUTING ->
CS_CONSUME_MEDIA
2011-06-24 13:29:46.945633 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996563750911 at 203.208.207.212 [BREAK]
2011-06-24 13:29:46.945633 [DEBUG] switch_core_state_machine.c:359
(sofia/external/03996563750911 at 203.208.207.212) State ROUTING going to sleep
2011-06-24 13:29:46.945633 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996563750911 at 203.208.207.212) Running State Change
CS_CONSUME_MEDIA
send 999 bytes to udp/[203.208.207.212]:5060 at 05:29:47.454251:
   ------------------------------------------------------------------------
   INVITE sip:03996563750911 at 203.208.207.212 SIP/2.0
   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKytFt9mX5HQNSj
   Max-Forwards: 70
   From: "" <sip:03996597632298 at 202.73.56.46>;tag=62eBXSt93tr7F
   To: <sip:03996563750911 at 203.208.207.212>
   Call-ID: d0f42376-18c5-122f-3387-0015c5fc7ad92011-06-24 13:29:46.945633
[DEBUG] switch_core_state_machine.c:378 (sofia/external/
03996563750911 at 203.208.207.212) State CONSUME_MEDIA

   CSeq: 14110629 INVITE
   Contact: <sip:mod_sofia at 202.73.56.46:5080>
2011-06-24 13:29:46.945633 [DEBUG] switch_core_state_machine.c:378
(sofia/external/03996563750911 at 203.208.207.212) State CONSUME_MEDIA going to
sleep
   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, PRACK, NOTIFY
   Supported: 100rel, timer, precondition, path, replaces
   Allow-Events: talk, hold, refer
   Privacy: none
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 204
   X-FS-Support: update_display
   P-Asserted-Identity: "" <sip:03996597632298 at 202.73.56.46>

   v=0
   o=FreeSWITCH 1308872756 1308872757 IN IP4 202.73.56.46
   s=FreeSWITCH
   c=IN IP4 202.73.56.46
   t=0 0
   m=audio 20630 RTP/AVP 18 3 101 13
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
2011-06-24 13:29:46.946974 [DEBUG] sofia.c:4646 Channel sofia/external/
03996563750911 at 203.208.207.212 entering state [calling][0]
recv 307 bytes from udp/[203.208.207.212]:5060 at 05:29:47.460017:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.1.1.46:5080
;rport=5080;branch=z9hG4bKytFt9mX5HQNSj;received=10.1.1.46
   From: "" <sip:03996597632298 at 10.1.1.46>;tag=62eBXSt93tr7F
   To: <sip:03996563750911 at 203.208.207.212>
   Call-ID: d0f42376-18c5-122f-3387-0015c5fc7ad9
   CSeq: 14110629 INVITE
   Content-Length:     0

   ------------------------------------------------------------------------
2011-06-24 13:29:46.960900 [DEBUG] switch_ivr.c:563 sofia/external/
03996597632298 at 203.208.207.212 Command Execute
playback(/usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav)
EXECUTE sofia/external/03996597632298 at 203.208.207.212playback(/usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav)
2011-06-24 13:29:46.960900 [DEBUG] switch_core_file.c:176 File
/usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav sample rate 11025
doesn't match requested rate 8000
2011-06-24 13:29:46.960900 [WARNING] switch_core_file.c:189 File has 2
channels, muxing to mono will occur.
2011-06-24 13:29:46.960900 [DEBUG] switch_ivr_play_say.c:1244 Codec
Activated L16 at 8000hz 2 channels 20ms
2011-06-24 13:29:46.981203 [INFO] mod_com_g729.c:119 ENCODER CREATE -
0x2aaab80894b8 0x5112110
recv 774 bytes from udp/[203.208.207.212]:5060 at 05:29:48.623941:
   ------------------------------------------------------------------------
   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP 10.1.1.46:5080
;rport=5080;branch=z9hG4bKytFt9mX5HQNSj;received=10.1.1.46
   From: "" <sip:03996597632298 at 10.1.1.46>;tag=62eBXSt93tr7F
   To: <sip:03996563750911 at 203.208.207.212
>;tag=2QGB951HCR30000E1D0001Ll000B8RV0KORLQK
   Call-ID: d0f42376-18c5-122f-3387-0015c5fc7ad9
   CSeq: 14110629 INVITE
   Contact: <sip:203.208.207.212:5060>
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
SUBSCRIBE, UPDATE
   Content-Type: application/sdp
   Require: 100rel
   RSeq: 1
   Content-Length:   231

   v=0
   o=- 508550174225779858 1 IN IP4 203.208.207.212
   s=session
   c=IN IP4 203.208.207.202
   t=0 0
   m=audio 25150 RTP/AVP 18 101
   a=rtpmap:18 G729/8000
   a=fmtp:18 annexb=no
   a=rtpmap:101 telephone-event/8000
   a=ptime:20
   a=sendrecv
   ------------------------------------------------------------------------
send 651 bytes to udp/[203.208.207.212]:5060 at 05:29:48.624297:
   ------------------------------------------------------------------------
   PRACK sip:203.208.207.212:5060 SIP/2.0
   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKZ38jBge9e0Bce
   Max-Forwards: 70
   From: "" <sip:03996597632298 at 202.73.56.46>;tag=62eBXSt93tr7F
   To: <sip:03996563750911 at 203.208.207.212
>;tag=2QGB951HCR30000E1D0001Ll000B8RV0KORLQK
   Call-ID: d0f42376-18c5-122f-3387-0015c5fc7ad9
   CSeq: 14110630 PRACK
   Contact: <sip:mod_sofia at 202.73.56.46:5080>
   RAck: 1 14110629 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, PRACK, NOTIFY
   Supported: 100rel, timer, precondition, path, replaces
   Content-Length: 0

   ------------------------------------------------------------------------
2011-06-24 13:29:48.115941 [INFO] sofia.c:729 sofia/external/
03996563750911 at 203.208.207.212 Update Callee ID to "Outbound Call"
<03996563750911>
2011-06-24 13:29:48.115941 [DEBUG] sofia.c:4646 Channel sofia/external/
03996563750911 at 203.208.207.212 entering state [proceeding][183]
2011-06-24 13:29:48.115941 [DEBUG] sofia.c:4657 Remote SDP:
v=0
o=- 508550174225779858 1 IN IP4 203.208.207.212
s=session
c=IN IP4 203.208.207.202
t=0 0
m=audio 25150 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=ptime:20

2011-06-24 13:29:48.115941 [DEBUG] sofia_glue.c:4467 Audio Codec Compare
[G729:18:8000:20:8000]/[G729:18:8000:20:8000]
2011-06-24 13:29:48.115941 [DEBUG] sofia_glue.c:2757 Set Codec
sofia/external/03996563750911 at 203.208.207.212 G729/8000 20 ms 160 samples
8000 bits
2011-06-24 13:29:48.115941 [DEBUG] sofia_glue.c:4565 Set 2833 dtmf send
payload to 101
2011-06-24 13:29:48.115941 [DEBUG] sofia_glue.c:2987 AUDIO RTP
[sofia/external/03996563750911 at 203.208.207.212] 10.1.1.46 port 20630 ->
203.208.207.202 port 25150 codec: 18 ms: 20
2011-06-24 13:29:48.115941 [DEBUG] switch_rtp.c:1607 Starting timer [soft]
160 bytes per 20ms
2011-06-24 13:29:48.119065 [DEBUG] sofia_glue.c:3228 Set 2833 dtmf send
payload to 101
2011-06-24 13:29:48.119065 [DEBUG] sofia_glue.c:3233 Set 2833 dtmf receive
payload to 101
2011-06-24 13:29:48.119065 [NOTICE] sofia_glue.c:3680 Pre-Answer
sofia/external/03996563750911 at 203.208.207.212!
2011-06-24 13:29:48.119065 [DEBUG] switch_channel.c:2627 (sofia/external/
03996563750911 at 203.208.207.212) Callstate Change RINGING -> EARLY
2011-06-24 13:29:48.120495 [DEBUG] switch_ivr_originate.c:3408 Originate
Resulted in Success: [sofia/external/03996563750911 at 203.208.207.212]
2011-06-24 13:29:48.120495 [DEBUG] mod_commands.c:3205 (sofia/external/
03996563750911 at 203.208.207.212) State Change CS_CONSUME_MEDIA -> CS_EXECUTE
2011-06-24 13:29:48.120495 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996563750911 at 203.208.207.212 [BREAK]
2011-06-24 13:29:48.120495 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996563750911 at 203.208.207.212) Running State Change
CS_EXECUTE
2011-06-24 13:29:48.120495 [DEBUG] switch_core_state_machine.c:366
(sofia/external/03996563750911 at 203.208.207.212) State EXECUTE
2011-06-24 13:29:48.120495 [DEBUG] mod_sofia.c:240 sofia/external/
03996563750911 at 203.208.207.212 SOFIA EXECUTE
2011-06-24 13:29:48.120495 [DEBUG] switch_core_state_machine.c:157
sofia/external/03996563750911 at 203.208.207.212 Standard EXECUTE
EXECUTE sofia/external/03996563750911 at 203.208.207.212 park()
2011-06-24 13:29:48.122782 [DEBUG] switch_core_session.c:954 Send signal
sofia/external/03996597632298 at 203.208.207.212 [BREAK]
recv 382 bytes from udp/[203.208.207.212]:5060 at 05:29:48.630447:
   ------------------------------------------------------------------------
   SIP/2.0 200 Ok
   Via: SIP/2.0/UDP 10.1.1.46:5080
;rport=5080;branch=z9hG4bKZ38jBge9e0Bce;received=10.1.1.46
   From: "" <sip:03996597632298 at 10.1.1.46>;tag=62eBXSt93tr7F
   To: <sip:03996563750911 at 203.208.207.212
>;tag=2QGB951HCR30000E1D0001Ll000B8RV0KORLQK
   Call-ID: d0f42376-18c5-122f-3387-0015c5fc7ad9
   CSeq: 14110630 PRACK
   Contact: <sip:203.208.207.212:5060>
   Content-Length:     0

   ------------------------------------------------------------------------
2011-06-24 13:29:48.122782 [DEBUG] switch_ivr.c:563 sofia/external/
03996597632298 at 203.208.207.212 Command Execute
playback(tone_stream://%(2000,4000,440,480);loops=10)
EXECUTE sofia/external/03996597632298 at 203.208.207.212playback(tone_stream://%(2000,4000,440,480);loops=10)
2011-06-24 13:29:48.124434 [DEBUG] switch_ivr_bridge.c:1480 (sofia/external/
03996597632298 at 203.208.207.212) State Change CS_EXECUTE -> CS_HIBERNATE
2011-06-24 13:29:48.124434 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996597632298 at 203.208.207.212 [BREAK]
2011-06-24 13:29:48.124434 [DEBUG] switch_ivr_bridge.c:1482 (sofia/external/
03996563750911 at 203.208.207.212) State Change CS_EXECUTE -> CS_HIBERNATE
2011-06-24 13:29:48.124434 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996563750911 at 203.208.207.212 [BREAK]
2011-06-24 13:29:48.124434 [DEBUG] switch_core_session.c:771 Send signal
sofia/external/03996563750911 at 203.208.207.212 [BREAK]
2011-06-24 13:29:48.124434 [DEBUG] switch_core_session.c:771 Send signal
sofia/external/03996597632298 at 203.208.207.212 [BREAK]
2011-06-24 13:29:48.124434 [DEBUG] switch_ivr_play_say.c:1244 Codec
Activated L16 at 8000hz 1 channels 20ms
2011-06-24 13:29:48.124434 [DEBUG] switch_ivr_play_say.c:1581 done playing
file
2011-06-24 13:29:48.124434 [DEBUG] switch_core_state_machine.c:366
(sofia/external/03996563750911 at 203.208.207.212) State EXECUTE going to sleep
2011-06-24 13:29:48.124434 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996563750911 at 203.208.207.212) Running State Change
CS_HIBERNATE
2011-06-24 13:29:48.124434 [DEBUG] switch_core_state_machine.c:381
(sofia/external/03996563750911 at 203.208.207.212) State HIBERNATE
2011-06-24 13:29:48.124434 [DEBUG] mod_sofia.c:221 sofia/external/
03996563750911 at 203.208.207.212 SOFIA HIBERNATE
2011-06-24 13:29:48.124434 [DEBUG] switch_ivr_bridge.c:731 (sofia/external/
03996563750911 at 203.208.207.212) State Change CS_HIBERNATE -> CS_RESET
2011-06-24 13:29:48.124434 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996563750911 at 203.208.207.212 [BREAK]
2011-06-24 13:29:48.124434 [DEBUG] switch_core_state_machine.c:381
(sofia/external/03996563750911 at 203.208.207.212) State HIBERNATE going to
sleep
2011-06-24 13:29:48.124434 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996563750911 at 203.208.207.212) Running State Change
CS_RESET
2011-06-24 13:29:48.124434 [DEBUG] switch_core_state_machine.c:362
(sofia/external/03996563750911 at 203.208.207.212) State RESET
2011-06-24 13:29:48.124434 [DEBUG] mod_sofia.c:165 sofia/external/
03996563750911 at 203.208.207.212 SOFIA RESET
2011-06-24 13:29:48.124434 [DEBUG] switch_ivr_bridge.c:716 sofia/external/
03996563750911 at 203.208.207.212 CUSTOM RESET
2011-06-24 13:29:48.124434 [DEBUG] switch_core_state_machine.c:66
sofia/external/03996563750911 at 203.208.207.212 Standard RESET
2011-06-24 13:29:48.124434 [DEBUG] switch_core_state_machine.c:362
(sofia/external/03996563750911 at 203.208.207.212) State RESET going to sleep
2011-06-24 13:29:48.126015 [DEBUG] switch_core_session.c:709 Send signal
sofia/external/03996563750911 at 203.208.207.212 [BREAK]
2011-06-24 13:29:48.142264 [DEBUG] switch_ivr_play_say.c:1581 done playing
file
2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:366
(sofia/external/03996597632298 at 203.208.207.212) State EXECUTE going to sleep
2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996597632298 at 203.208.207.212) Running State Change
CS_HIBERNATE
2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:381
(sofia/external/03996597632298 at 203.208.207.212) State HIBERNATE
2011-06-24 13:29:48.142264 [DEBUG] mod_sofia.c:221 sofia/external/
03996597632298 at 203.208.207.212 SOFIA HIBERNATE
2011-06-24 13:29:48.142264 [DEBUG] switch_ivr_bridge.c:731 (sofia/external/
03996597632298 at 203.208.207.212) State Change CS_HIBERNATE -> CS_RESET
2011-06-24 13:29:48.142264 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996597632298 at 203.208.207.212 [BREAK]
2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:381
(sofia/external/03996597632298 at 203.208.207.212) State HIBERNATE going to
sleep
2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996597632298 at 203.208.207.212) Running State Change
CS_RESET
2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:362
(sofia/external/03996597632298 at 203.208.207.212) State RESET
2011-06-24 13:29:48.142264 [DEBUG] mod_sofia.c:165 sofia/external/
03996597632298 at 203.208.207.212 SOFIA RESET
2011-06-24 13:29:48.142264 [DEBUG] switch_ivr_bridge.c:716 sofia/external/
03996597632298 at 203.208.207.212 CUSTOM RESET
2011-06-24 13:29:48.142264 [DEBUG] switch_ivr_bridge.c:723 (sofia/external/
03996597632298 at 203.208.207.212) State Change CS_RESET -> CS_SOFT_EXECUTE
2011-06-24 13:29:48.142264 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996597632298 at 203.208.207.212 [BREAK]
2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:362
(sofia/external/03996597632298 at 203.208.207.212) State RESET going to sleep
2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996597632298 at 203.208.207.212) Running State Change
CS_SOFT_EXECUTE
2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:372
(sofia/external/03996597632298 at 203.208.207.212) State SOFT_EXECUTE
2011-06-24 13:29:48.142264 [DEBUG] mod_sofia.c:558 SOFIA SOFT_EXECUTE
2011-06-24 13:29:48.142264 [DEBUG] switch_ivr_bridge.c:741 sofia/external/
03996597632298 at 203.208.207.212 CUSTOM SOFT_EXECUTE
2011-06-24 13:29:48.142264 [DEBUG] switch_ivr_bridge.c:761 (sofia/external/
03996563750911 at 203.208.207.212) State Change CS_RESET -> CS_SOFT_EXECUTE
2011-06-24 13:29:48.142264 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996563750911 at 203.208.207.212 [BREAK]
2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996563750911 at 203.208.207.212) Running State Change
CS_SOFT_EXECUTE
2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:372
(sofia/external/03996563750911 at 203.208.207.212) State SOFT_EXECUTE
2011-06-24 13:29:48.142264 [DEBUG] mod_sofia.c:558 SOFIA SOFT_EXECUTE
2011-06-24 13:29:48.142264 [DEBUG] switch_ivr_bridge.c:741 sofia/external/
03996563750911 at 203.208.207.212 CUSTOM SOFT_EXECUTE
2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:204
sofia/external/03996563750911 at 203.208.207.212 Standard SOFT_EXECUTE
2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:372
(sofia/external/03996563750911 at 203.208.207.212) State SOFT_EXECUTE going to
sleep
2011-06-24 13:29:48.161779 [DEBUG] switch_channel.c:2535 (sofia/external/
03996563750911 at 203.208.207.212) Callstate Change EARLY -> HANGUP
2011-06-24 13:29:48.161779 [NOTICE] switch_ivr_originate.c:1045 Hangup
sofia/external/03996563750911 at 203.208.207.212 [CS_SOFT_EXECUTE]
[DESTINATION_OUT_OF_ORDER]
2011-06-24 13:29:48.161779 [DEBUG] switch_channel.c:2551 Send signal
sofia/external/03996563750911 at 203.208.207.212 [KILL]
2011-06-24 13:29:48.161779 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996563750911 at 203.208.207.212 [BREAK]
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996563750911 at 203.208.207.212) Running State Change
CS_HANGUP
2011-06-24 13:29:48.161779 [DEBUG] switch_channel.c:2535 (sofia/external/
03996597632298 at 203.208.207.212) Callstate Change ACTIVE -> HANGUP
2011-06-24 13:29:48.161779 [NOTICE] switch_ivr_bridge.c:772 Hangup
sofia/external/03996597632298 at 203.208.207.212 [CS_SOFT_EXECUTE]
[ORIGINATOR_CANCEL]
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:557
(sofia/external/03996563750911 at 203.208.207.212) State HANGUP
2011-06-24 13:29:48.161779 [DEBUG] mod_sofia.c:457 Channel sofia/external/
03996563750911 at 203.208.207.212 hanging up, cause: DESTINATION_OUT_OF_ORDER
2011-06-24 13:29:48.161779 [DEBUG] mod_sofia.c:510 Sending CANCEL to
sofia/external/03996563750911 at 203.208.207.212
2011-06-24 13:29:48.161779 [DEBUG] switch_channel.c:2551 Send signal
sofia/external/03996597632298 at 203.208.207.212 [KILL]
2011-06-24 13:29:48.161779 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996597632298 at 203.208.207.212 [BREAK]
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:372
(sofia/external/03996597632298 at 203.208.207.212) State SOFT_EXECUTE going to
sleep
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996597632298 at 203.208.207.212) Running State Change
CS_HANGUP
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:46
sofia/external/03996563750911 at 203.208.207.212 Standard HANGUP, cause:
DESTINATION_OUT_OF_ORDER
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:557
(sofia/external/03996563750911 at 203.208.207.212) State HANGUP going to sleep
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:351
(sofia/external/03996563750911 at 203.208.207.212) State Change CS_HANGUP ->
CS_REPORTING
2011-06-24 13:29:48.161779 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996563750911 at 203.208.207.212 [BREAK]
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996563750911 at 203.208.207.212) Running State Change
CS_REPORTING
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:617
(sofia/external/03996563750911 at 203.208.207.212) State REPORTING
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:53
sofia/external/03996563750911 at 203.208.207.212 Standard REPORTING, cause:
DESTINATION_OUT_OF_ORDER
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:617
(sofia/external/03996563750911 at 203.208.207.212) State REPORTING going to
sleep
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:557
(sofia/external/03996597632298 at 203.208.207.212) State HANGUP
2011-06-24 13:29:48.161779 [DEBUG] mod_sofia.c:457 Channel sofia/external/
03996597632298 at 203.208.207.212 hanging up, cause: ORIGINATOR_CANCEL
2011-06-24 13:29:48.161779 [DEBUG] mod_sofia.c:500 Sending BYE to
sofia/external/03996597632298 at 203.208.207.212
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:345
(sofia/external/03996563750911 at 203.208.207.212) State Change CS_REPORTING ->
CS_DESTROY
2011-06-24 13:29:48.161779 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996563750911 at 203.208.207.212 [BREAK]
2011-06-24 13:29:48.161779 [DEBUG] switch_core_session.c:1288 Session 52
(sofia/external/03996563750911 at 203.208.207.212) Locked, Waiting on external
entities
2011-06-24 13:29:48.161779 [NOTICE] switch_core_session.c:1306 Session 52
(sofia/external/03996563750911 at 203.208.207.212) Ended
2011-06-24 13:29:48.161779 [NOTICE] switch_core_session.c:1308 Close Channel
sofia/external/03996563750911 at 203.208.207.212 [CS_DESTROY]
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:449
(sofia/external/03996563750911 at 203.208.207.212) Callstate Change HANGUP ->
DOWN
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:452
(sofia/external/03996563750911 at 203.208.207.212) Running State Change
CS_DESTROY
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:462
(sofia/external/03996563750911 at 203.208.207.212) State DESTROY
2011-06-24 13:29:48.161779 [DEBUG] mod_sofia.c:362 sofia/external/
03996563750911 at 203.208.207.212 SOFIA DESTROY
2011-06-24 13:29:48.161779 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
0x2aaaac03fff8 (nil)
2011-06-24 13:29:48.161779 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
0x2aaaac03fff8 (nil)
2011-06-24 13:29:48.161779 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
0x2aaaac068fc8 (nil)
2011-06-24 13:29:48.161779 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
0x2aaaac068fc8 (nil)
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:46
sofia/external/03996597632298 at 203.208.207.212 Standard HANGUP, cause:
ORIGINATOR_CANCEL
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:60
sofia/external/03996563750911 at 203.208.207.212 Standard DESTROY
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:557
(sofia/external/03996597632298 at 203.208.207.212) State HANGUP going to sleep
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:462
(sofia/external/03996563750911 at 203.208.207.212) State DESTROY going to sleep
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:351
(sofia/external/03996597632298 at 203.208.207.212) State Change CS_HANGUP ->
CS_REPORTING
2011-06-24 13:29:48.161779 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996597632298 at 203.208.207.212 [BREAK]
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:320
(sofia/external/03996597632298 at 203.208.207.212) Running State Change
CS_REPORTING
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:617
(sofia/external/03996597632298 at 203.208.207.212) State REPORTING
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:53
sofia/external/03996597632298 at 203.208.207.212 Standard REPORTING, cause:
ORIGINATOR_CANCEL
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:617
(sofia/external/03996597632298 at 203.208.207.212) State REPORTING going to
sleep
send 390 bytes to udp/[203.208.207.212]:5060 at 05:29:48.671783:
   ------------------------------------------------------------------------
   CANCEL sip:03996563750911 at 203.208.207.212 SIP/2.0
   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKytFt9mX5HQNSj
   Max-Forwards: 70
   From: "" <sip:03996597632298 at 202.73.56.46>;tag=62eBXSt93tr7F
   To: <sip:03996563750911 at 203.208.207.212>
   Call-ID: d0f42376-18c5-122f-3387-0015c5fc7ad9
   CSeq: 14110629 CANCEL
   Reason: Q.850;cause=27;text="DESTINATION_OUT_OF_ORDER"
   Content-Length: 0

   ------------------------------------------------------------------------
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:345
(sofia/external/03996597632298 at 203.208.207.212) State Change CS_REPORTING ->
CS_DESTROY
2011-06-24 13:29:48.161779 [DEBUG] switch_core_session.c:1116 Send signal
sofia/external/03996597632298 at 203.208.207.212 [BREAK]
2011-06-24 13:29:48.161779 [DEBUG] switch_core_session.c:1288 Session 51
(sofia/external/03996597632298 at 203.208.207.212) Locked, Waiting on external
entities
2011-06-24 13:29:48.161779 [NOTICE] switch_core_session.c:1306 Session 51
(sofia/external/03996597632298 at 203.208.207.212) Ended
2011-06-24 13:29:48.161779 [NOTICE] switch_core_session.c:1308 Close Channel
sofia/external/03996597632298 at 203.208.207.212 [CS_DESTROY]
send 677 bytes to udp/[203.208.207.212]:5060 at 05:29:48.671936:
   ------------------------------------------------------------------------
   BYE sip:203.208.207.212:5060 SIP/2.0
   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bK0c2BDBZcc91yS
   Max-Forwards: 70
   From: "" <sip:03996563750911 at 202.73.56.46>;tag=5SNjUy956H2mm
   To: <sip:03996597632298 at 203.208.207.212
>;tag=2QGB951HCR30000E1D0001Ll000B8RX0S6R1TX
   Call-ID: ca5d29b7-18c5-122f-3387-0015c5fc7ad9
   CSeq: 14110627 BYE
   Contact: <sip:mod_sofia at 202.73.56.46:5080>
   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, PRACK, NOTIFY
   Supported: 100rel, timer, precondition, path, replaces
   Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL"
   Content-Length: 0

   ------------------------------------------------------------------------
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:449
(sofia/external/03996597632298 at 203.208.207.212) Callstate Change HANGUP ->
DOWN
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:452
(sofia/external/03996597632298 at 203.208.207.212) Running State Change
CS_DESTROY
2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:462
(sofia/external/03996597632298 at 203.208.207.212) State DESTROY
2011-06-24 13:29:48.161779 [DEBUG] mod_sofia.c:362 sofia/external/
03996597632298 at 203.208.207.212 SOFIA DESTROY
2011-06-24 13:29:48.161779 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
0x2aaab8030878 (nil)
2011-06-24 13:29:48.161779 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
0x2aaab8030878 (nil)
2011-06-24 13:29:48.161779 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
0x2aaab80894b8 0x5112110
2011-06-24 13:29:48.161779 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
0x2aaab80894b8 (nil)
2011-06-24 13:29:48.166430 [INFO] mod_com_g729.c:83 ENCODER DESTROY -
0x2aaab80894b8 0x5112110
2011-06-24 13:29:48.166430 [DEBUG] switch_core_state_machine.c:60
sofia/external/03996597632298 at 203.208.207.212 Standard DESTROY
2011-06-24 13:29:48.166430 [DEBUG] switch_core_state_machine.c:462
(sofia/external/03996597632298 at 203.208.207.212) State DESTROY going to sleep
recv 383 bytes from udp/[203.208.207.212]:5060 at 05:29:48.679561:
   ------------------------------------------------------------------------
   SIP/2.0 200 Ok
   Via: SIP/2.0/UDP 10.1.1.46:5080
;rport=5080;branch=z9hG4bKytFt9mX5HQNSj;received=10.1.1.46
   From: "" <sip:03996597632298 at 10.1.1.46>;tag=62eBXSt93tr7F
   To: <sip:03996563750911 at 203.208.207.212
>;tag=2QGB951HCR30000E1D00000u000000013MCNME
   Call-ID: d0f42376-18c5-122f-3387-0015c5fc7ad9
   CSeq: 14110629 CANCEL
   Contact: <sip:203.208.207.212:5060>
   Content-Length:     0

   ------------------------------------------------------------------------
recv 731 bytes from udp/[203.208.207.212]:5060 at 05:29:48.681395:
   ------------------------------------------------------------------------
   UPDATE sip:mod_sofia at 10.1.1.46:5080 SIP/2.0
   Via: SIP/2.0/UDP 203.208.207.212:5060
;branch=z9hG4bK00151746C47A8307FCE25ED9D752
   From: <sip:03996563750911 at 203.208.207.212
>;tag=2QGB951HCR30000E1D0001Ll000B8RV0KORLQK
   To: "" <sip:03996597632298 at 10.1.1.46>;tag=62eBXSt93tr7F
   Call-ID: d0f42376-18c5-122f-3387-0015c5fc7ad9
   CSeq: 46496 UPDATE
   Contact: <sip:03996563750911 at 203.208.207.212:5060>
   Content-Type: application/sdp
   Max-Forwards: 70
   Supported: 100rel, timer, replaces
   Content-Length:   231

   v=0
   o=- 508550174225779858 2 IN IP4 203.208.207.212
   s=session
   c=IN IP4 203.208.207.202
   t=0 0
   m=audio 25150 RTP/AVP 18 101
   a=rtpmap:18 G729/8000
   a=fmtp:18 annexb=no
   a=rtpmap:101 telephone-event/8000
   a=ptime:20
   a=inactive
   ------------------------------------------------------------------------
send 895 bytes to udp/[203.208.207.212]:5060 at 05:29:48.681661:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 203.208.207.212:5060
;branch=z9hG4bK00151746C47A8307FCE25ED9D752
   From: <sip:03996563750911 at 203.208.207.212
>;tag=2QGB951HCR30000E1D0001Ll000B8RV0KORLQK
   To: "" <sip:03996597632298 at 10.1.1.46>;tag=62eBXSt93tr7F
   Call-ID: d0f42376-18c5-122f-3387-0015c5fc7ad9
   CSeq: 46496 UPDATE
   Contact: <sip:mod_sofia at 202.73.56.46:5080>
   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, PRACK, NOTIFY
   Supported: 100rel, timer, precondition, path, replaces
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 216

   v=0
   o=FreeSWITCH 1308872756 1308872758 IN IP4 202.73.56.46
   s=FreeSWITCH
   c=IN IP4 202.73.56.46
   t=0 0
   m=audio 20630 RTP/AVP 18 101 3 13
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=inactive
   a=ptime:20
   ------------------------------------------------------------------------
recv 411 bytes from udp/[203.208.207.212]:5060 at 05:29:48.681824:
   ------------------------------------------------------------------------
   SIP/2.0 487 Request Terminated
   Via: SIP/2.0/UDP 10.1.1.46:5080
;rport=5080;branch=z9hG4bKytFt9mX5HQNSj;received=10.1.1.46
   From: "" <sip:03996597632298 at 10.1.1.46>;tag=62eBXSt93tr7F
   To: <sip:03996563750911 at 203.208.207.212
>;tag=2QGB951HCR30000E1D0001Ll000B8RV0KORLQK
   Call-ID: d0f42376-18c5-122f-3387-0015c5fc7ad9
   CSeq: 14110629 INVITE
   Reason: SIP;cause=487;text="Request Terminated"
   Content-Length:     0

   ------------------------------------------------------------------------
send 371 bytes to udp/[203.208.207.212]:5060 at 05:29:48.681913:
   ------------------------------------------------------------------------
   ACK sip:03996563750911 at 203.208.207.212 SIP/2.0
   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKytFt9mX5HQNSj
   Max-Forwards: 70
   From: "" <sip:03996597632298 at 202.73.56.46>;tag=62eBXSt93tr7F
   To: <sip:03996563750911 at 203.208.207.212
>;tag=2QGB951HCR30000E1D0001Ll000B8RV0KORLQK
   Call-ID: d0f42376-18c5-122f-3387-0015c5fc7ad9
   CSeq: 14110629 ACK
   Content-Length: 0

   ------------------------------------------------------------------------
recv 380 bytes from udp/[203.208.207.212]:5060 at 05:29:48.683041:
   ------------------------------------------------------------------------
   SIP/2.0 200 Ok
   Via: SIP/2.0/UDP 10.1.1.46:5080
;rport=5080;branch=z9hG4bK0c2BDBZcc91yS;received=10.1.1.46
   From: "" <sip:03996563750911 at 10.1.1.46>;tag=5SNjUy956H2mm
   To: <sip:03996597632298 at 203.208.207.212
>;tag=2QGB951HCR30000E1D0001Ll000B8RX0S6R1TX
   Call-ID: ca5d29b7-18c5-122f-3387-0015c5fc7ad9
   CSeq: 14110627 BYE
   Contact: <sip:203.208.207.212:5060>
   Content-Length:     0

   ------------------------------------------------------------------------
2011-06-24 13:29:48.524051 [DEBUG] switch_scheduler.c:138 Deleting task 27
switch_ivr_schedule_hangup (c0bd700d-913c-42ad-b68f-81001bf658b8)






On Fri, Jun 24, 2011 at 2:41 PM, Michael Collins <msc at freeswitch.org> wrote:

> Pastebin the entire debug log, including the siptrace. Also include the
> originate line and any other dialplan config that might be used.
> -MC
>
>
> On Thu, Jun 23, 2011 at 11:26 PM, David Ma <mays.david at gmail.com> wrote:
>
>> Hi Michael,
>>
>> Unfortunately this problem still happens.
>>
>> I enabled "continue_on_fail" for leg-A when I originated the call. Leg-A
>> call went well. Then I originated leg-B call ("continue_on_fail" is NOT set
>> for leg-B), which failed for [DESTINATION_OUT_OF_ORDER]. As the consequence,
>> leg-A was hung up by FS automatically for [ORIGINATOR_CANCEL].
>>
>> The log excerpt follows.
>>
>> Do you think "continue_on_fail" should be also enabled for leg-B call?
>>
>> Thanks,
>> D.Ma
>>
>> 2011-06-24 13:29:35.887830 [DEBUG] switch_ivr_originate.c:1971 variable
>> string 5 = [continue_on_fail=true]
>>
>> 2011-06-24 13:29:35.887830 [NOTICE] switch_channel.c:808 New Channel
>> sofia/external/03996597632298 at 203.208.207.212[c0bd700d-913c-42ad-b68f-81001bf658b8]
>> [...]
>> 2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:372
>> (sofia/external/03996563750911 at 203.208.207.212) State SOFT_EXECUTE going
>> to sleep
>> 2011-06-24 13:29:48.161779 [DEBUG] switch_channel.c:2535 (sofia/external/
>> 03996563750911 at 203.208.207.212) Callstate Change EARLY -> HANGUP
>> 2011-06-24 13:29:48.161779 [NOTICE] switch_ivr_originate.c:1045 Hangup
>> sofia/external/03996563750911 at 203.208.207.212 [CS_SOFT_EXECUTE]
>> [DESTINATION_OUT_OF_ORDER]
>> 2011-06-24 13:29:48.161779 [DEBUG] switch_channel.c:2551 Send signal
>> sofia/external/03996563750911 at 203.208.207.212 [KILL]
>> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/external/03996563750911 at 203.208.207.212 [BREAK]
>> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:320
>> (sofia/external/03996563750911 at 203.208.207.212) Running State Change
>> CS_HANGUP
>> 2011-06-24 13:29:48.161779 [DEBUG] switch_channel.c:2535 (sofia/external/
>> 03996597632298 at 203.208.207.212) Callstate Change ACTIVE -> HANGUP
>> 2011-06-24 13:29:48.161779 [NOTICE] switch_ivr_bridge.c:772 Hangup
>> sofia/external/03996597632298 at 203.208.207.212 [CS_SOFT_EXECUTE]
>> [ORIGINATOR_CANCEL]
>> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:557
>> (sofia/external/03996563750911 at 203.208.207.212) State HANGUP
>> 2011-06-24 13:29:48.161779 [DEBUG] mod_sofia.c:457 Channel sofia/external/
>> 03996563750911 at 203.208.207.212 hanging up, cause:
>> DESTINATION_OUT_OF_ORDER
>> 2011-06-24 13:29:48.161779 [DEBUG] mod_sofia.c:510 Sending CANCEL to
>> sofia/external/03996563750911 at 203.208.207.212
>> 2011-06-24 13:29:48.161779 [DEBUG] switch_channel.c:2551 Send signal
>> sofia/external/03996597632298 at 203.208.207.212 [KILL]
>> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/external/03996597632298 at 203.208.207.212 [BREAK]
>>
>> On Fri, Jun 17, 2011 at 10:52 AM, David Ma <mays.david at gmail.com> wrote:
>>
>>> Hi Michael,
>>>
>>> Thanks very much for your prompt response! I appreciate the information
>>> provided.
>>>
>>> I was actually searching the the existence of such a variable. I was not
>>> so luck to find it out and thereby resort to the support forum.
>>>
>>> I've modified my code to build this parameter into my application. Will
>>> feedback to you after verification.
>>>
>>> Thanks again,
>>> D.Ma
>>>
>>> On Fri, Jun 17, 2011 at 4:51 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> How about setting this?
>>>> http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
>>>>
>>>> -MC
>>>>
>>>>
>>>> On Thu, Jun 16, 2011 at 1:32 AM, dma <mays.david at gmail.com> wrote:
>>>>
>>>>> I am creating a call-back solution. After leg-A answers, I originate
>>>>> leg-B
>>>>> call. After receiving SIP 183 from Leg-B, I bridge the 2 legs. However,
>>>>> in
>>>>> some cases, leg-A is automatically disconnected by FreeSwitch on leg-B
>>>>> failure, for example, DESTINATION_OUT_OF_ORDER. The application is not
>>>>> given
>>>>> a chance to handle leg-B failure event. This should not be a correct
>>>>> scenario because I never set "hangup-after-bridge", which is false by
>>>>> default.
>>>>>
>>>>> The right way should be, FreeSwitch doesn't hang up leg-A
>>>>> automatically, but
>>>>> give a chance for the application to decide what to do.
>>>>>
>>>>> Please see the logs below:
>>>>>
>>>>> =================================================
>>>>>
>>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
>>>>> string 0 = [origination_caller_id_number=03996563750914]
>>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
>>>>> string 1 = [originate_timeout=30]
>>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
>>>>> string 2 = [ccd_session_id=20110610105829676824]
>>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
>>>>> string 3 = [sip_cid_type=pid]
>>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971 variable
>>>>> string 4 = [privacy=yes]
>>>>> 2011-06-10 11:09:55.370506 [NOTICE] switch_channel.c:808 New Channel
>>>>> sofia/external/03996590031055 at 203.208.207.212
>>>>> [ea57b74b-a8c2-4fea-9683-98054dc03a79]
>>>>> 2011-06-10 11:09:55.370506 [DEBUG] mod_sofia.c:4129
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_NEW ->
>>>>> CS_INIT
>>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_core_session.c:1116 Send
>>>>> signal
>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_core_state_machine.c:320
>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>> CS_INIT
>>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_core_state_machine.c:356
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State INIT
>>>>> 2011-06-10 11:09:55.370506 [DEBUG] mod_sofia.c:84
>>>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA INIT
>>>>> send 984 bytes to udp/[203.208.207.212]:5060 at 03:09:55.477997:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>   INVITE sip:03996590031055 at 203.208.207.212 SIP/2.0
>>>>>   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKXjQ7eFpKypy5D
>>>>>   Max-Forwards: 70
>>>>>   From: "" &lt;sip:03996563750914 at 202.73.56.46&gt;;tag=H4ZN43UH9S9Qc
>>>>>   To: &lt;sip:03996590031055 at 203.208.207.212&gt;
>>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>>   CSeq: 13501633 INVITE
>>>>>   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
>>>>>   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
>>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>>> REGISTER, REFER, NOTIFY
>>>>>   Supported: timer, precondition, path, replaces
>>>>>   Allow-Events: talk, hold, refer
>>>>>   Privacy: none
>>>>>   Content-Type: application/sdp
>>>>>   Content-Disposition: session
>>>>>   Content-Length: 204
>>>>>   X-FS-Support: update_display
>>>>>   P-Asserted-Identity: "" &lt;sip:03996563750914 at 202.73.56.46&gt;
>>>>>
>>>>>   v=0
>>>>>   o=FreeSWITCH 1307645395 1307645396 IN IP4 202.73.56.46
>>>>>   s=FreeSWITCH
>>>>>   c=IN IP4 202.73.56.46
>>>>>   t=0 0
>>>>>   m=audio 30000 RTP/AVP 18 3 101 13
>>>>>   a=rtpmap:101 telephone-event/8000
>>>>>   a=fmtp:101 0-16
>>>>>   a=ptime:20
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> 2011-06-10 11:09:55.371674 [DEBUG] mod_sofia.c:124
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_INIT
>>>>> ->
>>>>> CS_ROUTING
>>>>> 2011-06-10 11:09:55.371674 [DEBUG] switch_core_session.c:1116 Send
>>>>> signal
>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>> 2011-06-10 11:09:55.371674 [DEBUG] sofia.c:4646 Channel
>>>>> sofia/external/03996590031055 at 203.208.207.212 entering state
>>>>> [calling][0]
>>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:356
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State INIT going to
>>>>> sleep
>>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:320
>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>> CS_ROUTING
>>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_channel.c:1657
>>>>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change DOWN
>>>>> ->
>>>>> RINGING
>>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:359
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State ROUTING
>>>>> 2011-06-10 11:09:55.373478 [DEBUG] mod_sofia.c:147
>>>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA ROUTING
>>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_ivr_originate.c:66
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change
>>>>> CS_ROUTING ->
>>>>> CS_CONSUME_MEDIA
>>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_session.c:1116 Send
>>>>> signal
>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:359
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State ROUTING going to
>>>>> sleep
>>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:320
>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>> CS_CONSUME_MEDIA
>>>>> 2011-06-10 11:09:55.374485 [DEBUG] switch_core_state_machine.c:378
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State CONSUME_MEDIA
>>>>> 2011-06-10 11:09:55.374485 [DEBUG] switch_core_state_machine.c:378
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State CONSUME_MEDIA
>>>>> going to
>>>>> sleep
>>>>> recv 307 bytes from udp/[203.208.207.212]:5060 at 03:09:55.485130:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>   SIP/2.0 100 Trying
>>>>>   Via: SIP/2.0/UDP
>>>>> 10.1.1.46:5080
>>>>> ;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>>>   To: &lt;sip:03996590031055 at 203.208.207.212&gt;
>>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>>   CSeq: 13501633 INVITE
>>>>>   Content-Length:     0
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> recv 669 bytes from udp/[203.208.207.212]:5060 at 03:09:56.677788:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>   SIP/2.0 183 Session Progress
>>>>>   Via: SIP/2.0/UDP
>>>>> 10.1.1.46:5080
>>>>> ;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>>>   To:
>>>>> &lt;sip:03996590031055 at 203.208.207.212
>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>>   CSeq: 13501633 INVITE
>>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
>>>>> SUBSCRIBE, UPDATE
>>>>>   Content-Type: application/sdp
>>>>>   Content-Length:   189
>>>>>
>>>>>   v=0
>>>>>   o=- 421265648 1 IN IP4 203.208.207.219
>>>>>   s=session
>>>>>   c=IN IP4 203.208.207.196
>>>>>   t=0 0
>>>>>   m=audio 30792 RTP/AVP 18 101
>>>>>   a=rtpmap:18 G729/8000
>>>>>   a=rtpmap:101 telephone-event/8000
>>>>>   a=sendrecv
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> 2011-06-10 11:09:56.571839 [INFO] sofia.c:729
>>>>> sofia/external/03996590031055 at 203.208.207.212 Update Callee ID to
>>>>> "Outbound
>>>>> Call" <03996590031055>
>>>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia.c:4646 Channel
>>>>> sofia/external/03996590031055 at 203.208.207.212 entering state
>>>>> [proceeding][183]
>>>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia.c:4657 Remote SDP:
>>>>> v=0
>>>>> o=- 421265648 1 IN IP4 203.208.207.219
>>>>> s=session
>>>>> c=IN IP4 203.208.207.196
>>>>> t=0 0
>>>>> m=audio 30792 RTP/AVP 18 101
>>>>> a=rtpmap:18 G729/8000
>>>>> a=rtpmap:101 telephone-event/8000
>>>>>
>>>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:4467 Audio Codec
>>>>> Compare
>>>>> [G729:18:8000:20:8000]/[G729:18:8000:20:8000]
>>>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:2757 Set Codec
>>>>> sofia/external/03996590031055 at 203.208.207.212 G729/8000 20 ms 160
>>>>> samples
>>>>> 8000 bits
>>>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:4565 Set 2833 dtmf send
>>>>> payload to 101
>>>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:2987 AUDIO RTP
>>>>> [sofia/external/03996590031055 at 203.208.207.212] 10.1.1.46 port 30000
>>>>> ->
>>>>> 203.208.207.196 port 30792 codec: 18 ms: 20
>>>>> 2011-06-10 11:09:56.571839 [DEBUG] switch_rtp.c:1607 Starting timer
>>>>> [soft]
>>>>> 160 bytes per 20ms
>>>>> 2011-06-10 11:09:56.573731 [DEBUG] sofia_glue.c:3228 Set 2833 dtmf send
>>>>> payload to 101
>>>>> 2011-06-10 11:09:56.573731 [DEBUG] sofia_glue.c:3233 Set 2833 dtmf
>>>>> receive
>>>>> payload to 101
>>>>> 2011-06-10 11:09:56.573731 [NOTICE] sofia_glue.c:3680 Pre-Answer
>>>>> sofia/external/03996590031055 at 203.208.207.212!
>>>>> 2011-06-10 11:09:56.573731 [DEBUG] switch_channel.c:2627
>>>>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change
>>>>> RINGING ->
>>>>> EARLY
>>>>> 2011-06-10 11:09:56.573731 [DEBUG] switch_ivr_originate.c:3408
>>>>> Originate
>>>>> Resulted in Success: [sofia/external/03996590031055 at 203.208.207.212]
>>>>> 2011-06-10 11:09:56.573731 [DEBUG] mod_commands.c:3205
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change
>>>>> CS_CONSUME_MEDIA -> CS_EXECUTE
>>>>> 2011-06-10 11:09:56.573731 [DEBUG] switch_core_session.c:1116 Send
>>>>> signal
>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>> 2011-06-10 11:09:56.575262 [DEBUG] switch_core_state_machine.c:320
>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>> CS_EXECUTE
>>>>> 2011-06-10 11:09:56.575262 [DEBUG] switch_core_state_machine.c:366
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State EXECUTE
>>>>> 2011-06-10 11:09:56.575262 [DEBUG] mod_sofia.c:240
>>>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA EXECUTE
>>>>> 2011-06-10 11:09:56.575262 [DEBUG] switch_core_state_machine.c:157
>>>>> sofia/external/03996590031055 at 203.208.207.212 Standard EXECUTE
>>>>> EXECUTE sofia/external/03996590031055 at 203.208.207.212 park()
>>>>> 2011-06-10 11:09:56.618770 [DEBUG] switch_rtp.c:2933 Correct ip/port
>>>>> confirmed.
>>>>> recv 669 bytes from udp/[203.208.207.212]:5060 at 03:10:01.562021:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>   SIP/2.0 183 Session Progress
>>>>>   Via: SIP/2.0/UDP
>>>>> 10.1.1.46:5080
>>>>> ;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>>>   To:
>>>>> &lt;sip:03996590031055 at 203.208.207.212
>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>>   CSeq: 13501633 INVITE
>>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
>>>>> SUBSCRIBE, UPDATE
>>>>>   Content-Type: application/sdp
>>>>>   Content-Length:   189
>>>>>
>>>>>   v=0
>>>>>   o=- 421265648 2 IN IP4 203.208.207.219
>>>>>   s=session
>>>>>   c=IN IP4 203.208.207.196
>>>>>   t=0 0
>>>>>   m=audio 30792 RTP/AVP 18 101
>>>>>   a=rtpmap:18 G729/8000
>>>>>   a=rtpmap:101 telephone-event/8000
>>>>>   a=sendrecv
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> 2011-06-10 11:10:01.455944 [INFO] sofia.c:729
>>>>> sofia/external/03996590031055 at 203.208.207.212 Update Callee ID to
>>>>> "03996590031055" <03996590031055>
>>>>> 2011-06-10 11:10:01.455944 [DEBUG] sofia.c:4641 Channel
>>>>> sofia/external/03996590031055 at 203.208.207.212 skipping state
>>>>> [proceeding][183]
>>>>> recv 669 bytes from udp/[203.208.207.212]:5060 at 03:10:01.563178:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>   SIP/2.0 183 Session Progress
>>>>>   Via: SIP/2.0/UDP
>>>>> 10.1.1.46:5080
>>>>> ;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>>>   To:
>>>>> &lt;sip:03996590031055 at 203.208.207.212
>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>>   CSeq: 13501633 INVITE
>>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
>>>>> SUBSCRIBE, UPDATE
>>>>>   Content-Type: application/sdp
>>>>>   Content-Length:   189
>>>>>
>>>>>   v=0
>>>>>   o=- 421265648 3 IN IP4 203.208.207.219
>>>>>   s=session
>>>>>   c=IN IP4 203.208.207.196
>>>>>   t=0 0
>>>>>   m=audio 30792 RTP/AVP 18 101
>>>>>   a=rtpmap:18 G729/8000
>>>>>   a=rtpmap:101 telephone-event/8000
>>>>>   a=sendrecv
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> 2011-06-10 11:10:01.457169 [DEBUG] sofia.c:4641 Channel
>>>>> sofia/external/03996590031055 at 203.208.207.212 skipping state
>>>>> [proceeding][183]
>>>>> recv 749 bytes from udp/[203.208.207.212]:5060 at 03:10:27.365284:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>   SIP/2.0 200 Ok
>>>>>   Via: SIP/2.0/UDP
>>>>> 10.1.1.46:5080
>>>>> ;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>>>   To:
>>>>> &lt;sip:03996590031055 at 203.208.207.212
>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>>   CSeq: 13501633 INVITE
>>>>>   Contact: <sip:203.208.207.212:5060>
>>>>>   Allow-Events: refer
>>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
>>>>> SUBSCRIBE, UPDATE
>>>>>   Content-Type: application/sdp
>>>>>   Supported: 100rel, timer, replaces
>>>>>   Content-Length:   189
>>>>>
>>>>>   v=0
>>>>>   o=- 421265648 4 IN IP4 203.208.207.219
>>>>>   s=session
>>>>>   c=IN IP4 203.208.207.196
>>>>>   t=0 0
>>>>>   m=audio 30792 RTP/AVP 18 101
>>>>>   a=rtpmap:18 G729/8000
>>>>>   a=rtpmap:101 telephone-event/8000
>>>>>   a=sendrecv
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> 2011-06-10 11:10:27.258092 [DEBUG] sofia.c:4646 Channel
>>>>> sofia/external/03996590031055 at 203.208.207.212 entering state
>>>>> [completing][200]
>>>>> 2011-06-10 11:10:27.258092 [DEBUG] sofia.c:4657 Remote SDP:
>>>>> v=0
>>>>> o=- 421265648 4 IN IP4 203.208.207.219
>>>>> s=session
>>>>> c=IN IP4 203.208.207.196
>>>>> t=0 0
>>>>> m=audio 30792 RTP/AVP 18 101
>>>>> a=rtpmap:18 G729/8000
>>>>> a=rtpmap:101 telephone-event/8000
>>>>>
>>>>> send 405 bytes to udp/[203.208.207.212]:5060 at 03:10:27.366562:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>   ACK sip:203.208.207.212:5060 SIP/2.0
>>>>>   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKyUg0ga7pUZmrS
>>>>>   Max-Forwards: 70
>>>>>   From: "" &lt;sip:03996563750914 at 202.73.56.46&gt;;tag=H4ZN43UH9S9Qc
>>>>>   To:
>>>>> &lt;sip:03996590031055 at 203.208.207.212
>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>>   CSeq: 13501633 ACK
>>>>>   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
>>>>>   Content-Length: 0
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> 2011-06-10 11:10:27.260267 [DEBUG] sofia.c:4646 Channel
>>>>> sofia/external/03996590031055 at 203.208.207.212 entering state
>>>>> [ready][200]
>>>>> 2011-06-10 11:10:27.260267 [DEBUG] switch_channel.c:2782
>>>>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change EARLY
>>>>> ->
>>>>> ACTIVE
>>>>> 2011-06-10 11:10:27.260267 [NOTICE] sofia.c:5175 Channel
>>>>> [sofia/external/03996590031055 at 203.208.207.212] has been answered
>>>>> 2011-06-10 11:10:27.264178 [DEBUG] switch_scheduler.c:214 Added task 23
>>>>> switch_ivr_schedule_hangup (ea57b74b-a8c2-4fea-9683-98054dc03a79) to
>>>>> run at
>>>>> 1307676927
>>>>> 2011-06-10 11:10:27.265202 [DEBUG] switch_core_session.c:954 Send
>>>>> signal
>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971 variable
>>>>> string 0 = [origination_caller_id_number=03996590031055]
>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971 variable
>>>>> string 1 = [ccd_session_id=20110610105829676824]
>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971 variable
>>>>> string 2 = [sip_cid_type=pid]
>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971 variable
>>>>> string 3 = [privacy=yes]
>>>>> 2011-06-10 11:10:27.266218 [NOTICE] switch_channel.c:808 New Channel
>>>>> sofia/external/03996563750914 at 203.208.207.212
>>>>> [30228d2b-756a-4a98-871d-db63a2955b52]
>>>>> 2011-06-10 11:10:27.266218 [DEBUG] mod_sofia.c:4129
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_NEW ->
>>>>> CS_INIT
>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_session.c:1116 Send
>>>>> signal
>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:320
>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>> CS_INIT
>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:356
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State INIT
>>>>> 2011-06-10 11:10:27.266218 [DEBUG] mod_sofia.c:84
>>>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA INIT
>>>>> 2011-06-10 11:10:27.266218 [DEBUG] mod_sofia.c:124
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_INIT
>>>>> ->
>>>>> CS_ROUTING
>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_session.c:1116 Send
>>>>> signal
>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:356
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State INIT going to
>>>>> sleep
>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:320
>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>> CS_ROUTING
>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_channel.c:1657
>>>>> (sofia/external/03996563750914 at 203.208.207.212) Callstate Change DOWN
>>>>> ->
>>>>> RINGING
>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:359
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State ROUTING
>>>>> send 984 bytes to udp/[203.208.207.212]:5060 at 03:10:27.373220:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>   INVITE sip:03996563750914 at 203.208.207.212 SIP/2.0
>>>>>   Via: SIP/2.0/UDP
>>>>> 202.73.56.46:5080;rport;branch=z9hG4bKZ49rj5Qtr8aBN2011-06-10
>>>>> 11:10:27.266218 [DEBUG] mod_sofia.c:147
>>>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA ROUTING
>>>>>
>>>>>   Max-Forwards: 70
>>>>>   From: "" &lt;sip:03996590031055 at 202.73.56.46&gt;;tag=jDSe6ycN62Zar
>>>>>   To: &lt;sip:03996563750914 at 203.208.207.212&gt;
>>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>>   CSeq: 13501649 INVITE
>>>>>   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
>>>>>   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
>>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>>> REGISTER, REFER, NOTIFY
>>>>>   Supported: timer, precondition, path, replaces
>>>>>   Allow-Events: talk, hold, refer
>>>>>   Privacy: none
>>>>>   Content-Type: application/sdp
>>>>>   Content-Disposition: session
>>>>>   Content-Length: 204
>>>>>   X-FS-Support: update_display
>>>>>   P-Asserted-Identity: "" &lt;sip:03996590031055 at 202.73.56.46&gt;
>>>>>
>>>>>   v=0
>>>>>   o=FreeSWITCH 1307646863 1307646864 IN IP4 202.73.56.46
>>>>>   s=FreeSWITCH
>>>>>   c=IN IP4 202.73.56.46
>>>>>   t=0 0
>>>>>   m=audio 28564 RTP/AVP 18 3 101 13
>>>>>   a=rtpmap:101 telephone-event/8000
>>>>>   a=fmtp:101 0-16
>>>>>   a=ptime:20
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> 2011-06-10 11:10:27.267630 [DEBUG] sofia.c:4646 Channel
>>>>> sofia/external/03996563750914 at 203.208.207.212 entering state
>>>>> [calling][0]
>>>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_ivr_originate.c:66
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change
>>>>> CS_ROUTING ->
>>>>> CS_CONSUME_MEDIA
>>>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_session.c:1116 Send
>>>>> signal
>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:359
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State ROUTING going to
>>>>> sleep
>>>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:320
>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>> CS_CONSUME_MEDIA
>>>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:378
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State CONSUME_MEDIA
>>>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:378
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State CONSUME_MEDIA
>>>>> going to
>>>>> sleep
>>>>> recv 307 bytes from udp/[203.208.207.212]:5060 at 03:10:27.378710:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>   SIP/2.0 100 Trying
>>>>>   Via: SIP/2.0/UDP
>>>>> 10.1.1.46:5080
>>>>> ;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>>>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>>>>   To: &lt;sip:03996563750914 at 203.208.207.212&gt;
>>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>>   CSeq: 13501649 INVITE
>>>>>   Content-Length:     0
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> 2011-06-10 11:10:27.278484 [DEBUG] switch_ivr.c:563
>>>>> sofia/external/03996590031055 at 203.208.207.212 Command Execute
>>>>> playback(/usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav)
>>>>> EXECUTE sofia/external/03996590031055 at 203.208.207.212
>>>>> playback(/usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav)
>>>>> 2011-06-10 11:10:27.278484 [DEBUG] switch_core_file.c:176 File
>>>>> /usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav sample rate
>>>>> 11025
>>>>> doesn't match requested rate 8000
>>>>> 2011-06-10 11:10:27.278484 [WARNING] switch_core_file.c:189 File has 2
>>>>> channels, muxing to mono will occur.
>>>>> 2011-06-10 11:10:27.278484 [DEBUG] switch_ivr_play_say.c:1244 Codec
>>>>> Activated L16 at 8000hz 2 channels 20ms
>>>>> 2011-06-10 11:10:27.298764 [INFO] mod_com_g729.c:119 ENCODER CREATE -
>>>>> 0x2aaab00310c0 0x2aaab00b20c0
>>>>> recv 666 bytes from udp/[203.208.207.212]:5060 at 03:10:28.618472:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>   SIP/2.0 183 Session Progress
>>>>>   Via: SIP/2.0/UDP
>>>>> 10.1.1.46:5080
>>>>> ;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>>>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>>>>   To:
>>>>> &lt;sip:03996563750914 at 203.208.207.212
>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
>>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>>   CSeq: 13501649 INVITE
>>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
>>>>> SUBSCRIBE, UPDATE
>>>>>   Content-Type: application/sdp
>>>>>   Content-Length:   186
>>>>>
>>>>>   v=0
>>>>>   o=- 131082 1 IN IP4 203.208.207.218
>>>>>   s=session
>>>>>   c=IN IP4 203.208.207.195
>>>>>   t=0 0
>>>>>   m=audio 45002 RTP/AVP 18 101
>>>>>   a=rtpmap:18 G729/8000
>>>>>   a=rtpmap:101 telephone-event/8000
>>>>>   a=sendrecv
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> 2011-06-10 11:10:28.513195 [INFO] sofia.c:729
>>>>> sofia/external/03996563750914 at 203.208.207.212 Update Callee ID to
>>>>> "Outbound
>>>>> Call" <03996563750914>
>>>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia.c:4646 Channel
>>>>> sofia/external/03996563750914 at 203.208.207.212 entering state
>>>>> [proceeding][183]
>>>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia.c:4657 Remote SDP:
>>>>> v=0
>>>>> o=- 131082 1 IN IP4 203.208.207.218
>>>>> s=session
>>>>> c=IN IP4 203.208.207.195
>>>>> t=0 0
>>>>> m=audio 45002 RTP/AVP 18 101
>>>>> a=rtpmap:18 G729/8000
>>>>> a=rtpmap:101 telephone-event/8000
>>>>>
>>>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:4467 Audio Codec
>>>>> Compare
>>>>> [G729:18:8000:20:8000]/[G729:18:8000:20:8000]
>>>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:2757 Set Codec
>>>>> sofia/external/03996563750914 at 203.208.207.212 G729/8000 20 ms 160
>>>>> samples
>>>>> 8000 bits
>>>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:4565 Set 2833 dtmf send
>>>>> payload to 101
>>>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:2987 AUDIO RTP
>>>>> [sofia/external/03996563750914 at 203.208.207.212] 10.1.1.46 port 28564
>>>>> ->
>>>>> 203.208.207.195 port 45002 codec: 18 ms: 20
>>>>> 2011-06-10 11:10:28.513195 [DEBUG] switch_rtp.c:1607 Starting timer
>>>>> [soft]
>>>>> 160 bytes per 20ms
>>>>> 2011-06-10 11:10:28.514938 [DEBUG] sofia_glue.c:3228 Set 2833 dtmf send
>>>>> payload to 101
>>>>> 2011-06-10 11:10:28.514938 [DEBUG] sofia_glue.c:3233 Set 2833 dtmf
>>>>> receive
>>>>> payload to 101
>>>>> 2011-06-10 11:10:28.514938 [NOTICE] sofia_glue.c:3680 Pre-Answer
>>>>> sofia/external/03996563750914 at 203.208.207.212!
>>>>> 2011-06-10 11:10:28.514938 [DEBUG] switch_channel.c:2627
>>>>> (sofia/external/03996563750914 at 203.208.207.212) Callstate Change
>>>>> RINGING ->
>>>>> EARLY
>>>>> 2011-06-10 11:10:28.514938 [DEBUG] switch_ivr_originate.c:3408
>>>>> Originate
>>>>> Resulted in Success: [sofia/external/03996563750914 at 203.208.207.212]
>>>>> 2011-06-10 11:10:28.514938 [DEBUG] mod_commands.c:3205
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change
>>>>> CS_CONSUME_MEDIA -> CS_EXECUTE
>>>>> 2011-06-10 11:10:28.514938 [DEBUG] switch_core_session.c:1116 Send
>>>>> signal
>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>> 2011-06-10 11:10:28.515884 [DEBUG] switch_core_state_machine.c:320
>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>> CS_EXECUTE
>>>>> 2011-06-10 11:10:28.515884 [DEBUG] switch_core_state_machine.c:366
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State EXECUTE
>>>>> 2011-06-10 11:10:28.515884 [DEBUG] mod_sofia.c:240
>>>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA EXECUTE
>>>>> 2011-06-10 11:10:28.515884 [DEBUG] switch_core_state_machine.c:157
>>>>> sofia/external/03996563750914 at 203.208.207.212 Standard EXECUTE
>>>>> EXECUTE sofia/external/03996563750914 at 203.208.207.212 park()
>>>>> 2011-06-10 11:10:28.516887 [DEBUG] switch_core_session.c:954 Send
>>>>> signal
>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_ivr_bridge.c:1480
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change
>>>>> CS_EXECUTE ->
>>>>> CS_HIBERNATE
>>>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:1116 Send
>>>>> signal
>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_ivr_bridge.c:1482
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change
>>>>> CS_EXECUTE ->
>>>>> CS_HIBERNATE
>>>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:1116 Send
>>>>> signal
>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:771 Send
>>>>> signal
>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:771 Send
>>>>> signal
>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_ivr_play_say.c:1581 done
>>>>> playing
>>>>> file
>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_ivr.c:563
>>>>> sofia/external/03996590031055 at 203.208.207.212 Command Execute
>>>>> playback(tone_stream://%(2000,4000,440,480);loops=10)
>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:366
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State EXECUTE going to
>>>>> sleep
>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:320
>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>> CS_HIBERNATE
>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:381
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State HIBERNATE
>>>>> 2011-06-10 11:10:28.519223 [DEBUG] mod_sofia.c:221
>>>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA HIBERNATE
>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_ivr_bridge.c:731
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change
>>>>> CS_HIBERNATE ->
>>>>> CS_RESET
>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_session.c:1116 Send
>>>>> signal
>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:381
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State HIBERNATE going
>>>>> to
>>>>> sleep
>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:320
>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>> CS_RESET
>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:362
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State RESET
>>>>> 2011-06-10 11:10:28.519223 [DEBUG] mod_sofia.c:165
>>>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA RESET
>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_ivr_bridge.c:716
>>>>> sofia/external/03996563750914 at 203.208.207.212 CUSTOM RESET
>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:66
>>>>> sofia/external/03996563750914 at 203.208.207.212 Standard RESET
>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:362
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State RESET going to
>>>>> sleep
>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_session.c:709 Send
>>>>> signal
>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>> recv 666 bytes from udp/[203.208.207.212]:5060 at 03:10:30.540814:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>   SIP/2.0 183 Session Progress
>>>>>   Via: SIP/2.0/UDP
>>>>> 10.1.1.46:5080
>>>>> ;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>>>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>>>>   To:
>>>>> &lt;sip:03996563750914 at 203.208.207.212
>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
>>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>>   CSeq: 13501649 INVITE
>>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
>>>>> SUBSCRIBE, UPDATE
>>>>>   Content-Type: application/sdp
>>>>>   Content-Length:   186
>>>>>
>>>>>   v=0
>>>>>   o=- 131082 2 IN IP4 203.208.207.218
>>>>>   s=session
>>>>>   c=IN IP4 203.208.207.195
>>>>>   t=0 0
>>>>>   m=audio 45002 RTP/AVP 18 101
>>>>>   a=rtpmap:18 G729/8000
>>>>>   a=rtpmap:101 telephone-event/8000
>>>>>   a=sendrecv
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> 2011-06-10 11:10:30.435309 [INFO] sofia.c:729
>>>>> sofia/external/03996563750914 at 203.208.207.212 Update Callee ID to
>>>>> "03996563750914" <03996563750914>
>>>>> 2011-06-10 11:10:30.435309 [DEBUG] sofia.c:4641 Channel
>>>>> sofia/external/03996563750914 at 203.208.207.212 skipping state
>>>>> [proceeding][183]
>>>>> 2011-06-10 11:10:30.691404 [WARNING] switch_core_session.c:1940 Cannot
>>>>> execute app 'playback' media required on an outbound channel that does
>>>>> not
>>>>> have media established
>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:366
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State EXECUTE going to
>>>>> sleep
>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>> CS_HIBERNATE
>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:381
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State HIBERNATE
>>>>> 2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:221
>>>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA HIBERNATE
>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:731
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change
>>>>> CS_HIBERNATE ->
>>>>> CS_RESET
>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_session.c:1116 Send
>>>>> signal
>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:381
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State HIBERNATE going
>>>>> to
>>>>> sleep
>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>> CS_RESET
>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:362
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State RESET
>>>>> 2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:165
>>>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA RESET
>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:716
>>>>> sofia/external/03996590031055 at 203.208.207.212 CUSTOM RESET
>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:723
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_RESET
>>>>> ->
>>>>> CS_SOFT_EXECUTE
>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_session.c:1116 Send
>>>>> signal
>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:362
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State RESET going to
>>>>> sleep
>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>> CS_SOFT_EXECUTE
>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:372
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State SOFT_EXECUTE
>>>>> 2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:558 SOFIA SOFT_EXECUTE
>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:741
>>>>> sofia/external/03996590031055 at 203.208.207.212 CUSTOM SOFT_EXECUTE
>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:761
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_RESET
>>>>> ->
>>>>> CS_SOFT_EXECUTE
>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_session.c:1116 Send
>>>>> signal
>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>> CS_SOFT_EXECUTE
>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:372
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State SOFT_EXECUTE
>>>>> 2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:558 SOFIA SOFT_EXECUTE
>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:741
>>>>> sofia/external/03996563750914 at 203.208.207.212 CUSTOM SOFT_EXECUTE
>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:204
>>>>> sofia/external/03996563750914 at 203.208.207.212 Standard SOFT_EXECUTE
>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:372
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State SOFT_EXECUTE
>>>>> going to
>>>>> sleep
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2535
>>>>> (sofia/external/03996563750914 at 203.208.207.212) Callstate Change EARLY
>>>>> ->
>>>>> HANGUP
>>>>> 2011-06-10 11:10:30.711798 [NOTICE] switch_ivr_originate.c:1045 Hangup
>>>>> sofia/external/03996563750914 at 203.208.207.212 [CS_SOFT_EXECUTE]
>>>>> [DESTINATION_OUT_OF_ORDER]
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2551 Send signal
>>>>> sofia/external/03996563750914 at 203.208.207.212 [KILL]
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send
>>>>> signal
>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>> CS_HANGUP
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2535
>>>>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change
>>>>> ACTIVE ->
>>>>> HANGUP
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State HANGUP
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:457 Channel
>>>>> sofia/external/03996563750914 at 203.208.207.212 hanging up, cause:
>>>>> DESTINATION_OUT_OF_ORDER
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:510 Sending CANCEL to
>>>>> sofia/external/03996563750914 at 203.208.207.212
>>>>> send 390 bytes to udp/[203.208.207.212]:5060 at 03:10:30.818391:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>   CANCEL sip:03996563750914 at 203.208.207.212 SIP/2.0
>>>>>   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKZ49rj5Qtr8aBN
>>>>>   Max-Forwards: 70
>>>>>   From: "" &lt;sip:03996590031055 at 202.73.56.46&gt;;tag=jDSe6ycN62Zar
>>>>>   To: &lt;sip:03996563750914 at 203.208.207.212&gt;
>>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>>   CSeq: 13501649 CANCEL
>>>>>   Reason: Q.850;cause=27;text="DESTINATION_OUT_OF_ORDER"
>>>>>   Content-Length: 0
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> 2011-06-10 11:10:30.711798 [NOTICE] switch_ivr_bridge.c:772 Hangup
>>>>> sofia/external/03996590031055 at 203.208.207.212 [CS_SOFT_EXECUTE]
>>>>> [ORIGINATOR_CANCEL]
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:46
>>>>> sofia/external/03996563750914 at 203.208.207.212 Standard HANGUP, cause:
>>>>> DESTINATION_OUT_OF_ORDER
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State HANGUP going to
>>>>> sleep
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:351
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_HANGUP
>>>>> ->
>>>>> CS_REPORTING
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send
>>>>> signal
>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>> CS_REPORTING
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:617
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State REPORTING
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:53
>>>>> sofia/external/03996563750914 at 203.208.207.212 Standard REPORTING,
>>>>> cause:
>>>>> DESTINATION_OUT_OF_ORDER
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:617
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State REPORTING going
>>>>> to
>>>>> sleep
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2551 Send signal
>>>>> sofia/external/03996590031055 at 203.208.207.212 [KILL]
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send
>>>>> signal
>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:372
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State SOFT_EXECUTE
>>>>> going to
>>>>> sleep
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>> CS_HANGUP
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:345
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change
>>>>> CS_REPORTING ->
>>>>> CS_DESTROY
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send
>>>>> signal
>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1288 Session
>>>>> 40
>>>>> (sofia/external/03996563750914 at 203.208.207.212) Locked, Waiting on
>>>>> external
>>>>> entities
>>>>> 2011-06-10 11:10:30.711798 [NOTICE] switch_core_session.c:1306 Session
>>>>> 40
>>>>> (sofia/external/03996563750914 at 203.208.207.212) Ended
>>>>> 2011-06-10 11:10:30.711798 [NOTICE] switch_core_session.c:1308 Close
>>>>> Channel
>>>>> sofia/external/03996563750914 at 203.208.207.212 [CS_DESTROY]
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:449
>>>>> (sofia/external/03996563750914 at 203.208.207.212) Callstate Change
>>>>> HANGUP ->
>>>>> DOWN
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:452
>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>> CS_DESTROY
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:462
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State DESTROY
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:362
>>>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA DESTROY
>>>>> 2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
>>>>> 0x2aaaac013028 (nil)
>>>>> 2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
>>>>> 0x2aaaac013028 (nil)
>>>>> 2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
>>>>> 0x2aaaac013088 (nil)
>>>>> 2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
>>>>> 0x2aaaac013088 (nil)
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:60
>>>>> sofia/external/03996563750914 at 203.208.207.212 Standard DESTROY
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:462
>>>>> (sofia/external/03996563750914 at 203.208.207.212) State DESTROY going to
>>>>> sleep
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State HANGUP
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:457 Channel
>>>>> sofia/external/03996590031055 at 203.208.207.212 hanging up, cause:
>>>>> ORIGINATOR_CANCEL
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:500 Sending BYE to
>>>>> sofia/external/03996590031055 at 203.208.207.212
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:46
>>>>> sofia/external/03996590031055 at 203.208.207.212 Standard HANGUP, cause:
>>>>> ORIGINATOR_CANCEL
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State HANGUP going to
>>>>> sleep
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:351
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_HANGUP
>>>>> ->
>>>>> CS_REPORTING
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send
>>>>> signal
>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>> CS_REPORTING
>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:617
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State REPORTING
>>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:53
>>>>> sofia/external/03996590031055 at 203.208.207.212 Standard REPORTING,
>>>>> cause:
>>>>> ORIGINATOR_CANCEL
>>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:617
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State REPORTING going
>>>>> to
>>>>> sleep
>>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:345
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change
>>>>> CS_REPORTING ->
>>>>> CS_DESTROY
>>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_session.c:1116 Send
>>>>> signal
>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_session.c:1288 Session
>>>>> 39
>>>>> (sofia/external/03996590031055 at 203.208.207.212) Locked, Waiting on
>>>>> external
>>>>> entities
>>>>> 2011-06-10 11:10:30.714001 [NOTICE] switch_core_session.c:1306 Session
>>>>> 39
>>>>> (sofia/external/03996590031055 at 203.208.207.212) Ended
>>>>> 2011-06-10 11:10:30.714001 [NOTICE] switch_core_session.c:1308 Close
>>>>> Channel
>>>>> sofia/external/03996590031055 at 203.208.207.212 [CS_DESTROY]
>>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:449
>>>>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change
>>>>> HANGUP ->
>>>>> DOWN
>>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:452
>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>> CS_DESTROY
>>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:462
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State DESTROY
>>>>> 2011-06-10 11:10:30.714001 [DEBUG] mod_sofia.c:362
>>>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA DESTROY
>>>>> 2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
>>>>> 0x2aaab0031060 (nil)
>>>>> 2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
>>>>> 0x2aaab0031060 (nil)
>>>>> 2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
>>>>> 0x2aaab00310c0 0x2aaab00b20c0
>>>>> 2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
>>>>> 0x2aaab00310c0 (nil)
>>>>> send 662 bytes to udp/[203.208.207.212]:5060 at 03:10:30.820530:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>   BYE sip:203.208.207.212:5060 SIP/2.0
>>>>>   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bK0D3Hm08XNH1Xg
>>>>>   Max-Forwards: 70
>>>>>   From: "" &lt;sip:03996563750914 at 202.73.56.46&gt;;tag=H4ZN43UH9S9Qc
>>>>>   To:
>>>>> &lt;sip:03996590031055 at 203.208.207.212
>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>>   CSeq: 13501634 BYE
>>>>>   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
>>>>>   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
>>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>>> REGISTER, REFER, NOTIFY
>>>>>   Supported: timer, precondition, path, replaces
>>>>>   Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL"
>>>>>   Content-Length: 0
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> 2011-06-10 11:10:30.715878 [INFO] mod_com_g729.c:83 ENCODER DESTROY -
>>>>> 0x2aaab00310c0 0x2aaab00b20c0
>>>>> 2011-06-10 11:10:30.715878 [DEBUG] switch_core_state_machine.c:60
>>>>> sofia/external/03996590031055 at 203.208.207.212 Standard DESTROY
>>>>> 2011-06-10 11:10:30.715878 [DEBUG] switch_core_state_machine.c:462
>>>>> (sofia/external/03996590031055 at 203.208.207.212) State DESTROY going to
>>>>> sleep
>>>>> recv 383 bytes from udp/[203.208.207.212]:5060 at 03:10:30.823302:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>   SIP/2.0 200 Ok
>>>>>   Via: SIP/2.0/UDP
>>>>> 10.1.1.46:5080
>>>>> ;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>>>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>>>>   To:
>>>>> &lt;sip:03996563750914 at 203.208.207.212
>>>>> &gt;;tag=2QGB951HCR30000E1D00000u00000001QXU3LU
>>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>>   CSeq: 13501649 CANCEL
>>>>>   Contact: <sip:203.208.207.212:5060>
>>>>>   Content-Length:     0
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> recv 411 bytes from udp/[203.208.207.212]:5060 at 03:10:30.824765:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>   SIP/2.0 487 Request Terminated
>>>>>   Via: SIP/2.0/UDP
>>>>> 10.1.1.46:5080
>>>>> ;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>>>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>>>>   To:
>>>>> &lt;sip:03996563750914 at 203.208.207.212
>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
>>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>>   CSeq: 13501649 INVITE
>>>>>   Reason: SIP;cause=487;text="Request Terminated"
>>>>>   Content-Length:     0
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> send 371 bytes to udp/[203.208.207.212]:5060 at 03:10:30.824891:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>   ACK sip:03996563750914 at 203.208.207.212 SIP/2.0
>>>>>   Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKZ49rj5Qtr8aBN
>>>>>   Max-Forwards: 70
>>>>>   From: "" &lt;sip:03996590031055 at 202.73.56.46&gt;;tag=jDSe6ycN62Zar
>>>>>   To:
>>>>> &lt;sip:03996563750914 at 203.208.207.212
>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
>>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>>   CSeq: 13501649 ACK
>>>>>   Content-Length: 0
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> recv 380 bytes from udp/[203.208.207.212]:5060 at 03:10:30.826375:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>   SIP/2.0 200 Ok
>>>>>   Via: SIP/2.0/UDP
>>>>> 10.1.1.46:5080
>>>>> ;rport=5080;branch=z9hG4bK0D3Hm08XNH1Xg;received=10.1.1.46
>>>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>>>   To:
>>>>> &lt;sip:03996590031055 at 203.208.207.212
>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>>   CSeq: 13501634 BYE
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> View this message in context:
>>>>> http://freeswitch-users.2379917.n2.nabble.com/Leg-A-is-automatically-disconnected-on-Leg-B-orginate-failure-tp6482192p6482192.html
>>>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bryansmart at bryansmart.com  Mon Jun 27 03:11:07 2011
From: bryansmart at bryansmart.com (Bryan Smart)
Date: Sun, 26 Jun 2011 19:11:07 -0400
Subject: [Freeswitch-users] Proper prompt gain/level
In-Reply-To: <0D3AEFF37C1643B79CE3F7A7EE9EAEC3@dell9400>
References: <FE081E27-8D62-4715-B7AA-64432598E013@bryansmart.com>
	<0D3AEFF37C1643B79CE3F7A7EE9EAEC3@dell9400>
Message-ID: <66E81300-46E9-4C81-B2E0-123BCADDCAEB@bryansmart.com>

I have tools to batch-process audio files. I just was not sure that regaining all of the prompt files was the best approach. I figured that the gain must have been reduced so dramatically for some sort of reason (to avoid clipping in some situation, to work better with the internal resampling, etc).

What AGC do you mean? I know that AGC has recently been added to conferencing, but the level of the prompts is a system-wide situation. As far as I know, there isn't AGC that can be applied on every channel, and, even if there was, there would surely be a processing hit, so the goal would be to avoid needing it, right?

The root problem, at least for me, is this. I need to add voice prompts and other audio for an IVR. I can't simply normalize all of my prompts to 0DB, as, even though they don't distort, they're so loud when compared to the stock prompts, they'll blow the phone out of my hand. To match them to the stock prompts, I must normalize them to around -16DB. I can do that, but it seems very wrong. At -16DB, nearly 85% of the potential gain of the channel is lost.

Try this... With the demo IVR (5000), add this before the sleep command in the dialplan.

<action application="set_audio_level" data="write 4"/>

That is the max gain boost available for a channel. The prompts should be really clipping with that much amplification, but they don't clip at all. At -16DB, you could literally amplify them to 6 times their native level without distorting. Native level is too low. Once I realized this, it became clear to me why Freeswitch sounded more quiet than Asterisk, at least when working with recorded prompts.

I suppose I could use set_audio_level on every last call, but I'm sure that real-time amplification, like AGC, is another processor drain that builds up with lots of calls. Besides, it seems weird to dramatically reduce the level of audio, and then waste cycles amplifying it back up in real-time.

Bryan

On Jun 26, 2011, at 5:20 PM, Jan Berger wrote:

> Wavepad is free and will do bulk changes to sound files.
> 
> Won't AGC solve this?
> 
> Jan
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bryan
> Smart
> Sent: 26. juni 2011 22:32
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Proper prompt gain/level
> 
> As part of creating prompts for my IVRs, I've tried to match the audio gain
> of my prompts with the gain of the stock English prompts. During this
> process, I noticed that the English Callie prompts are recorded extremely
> low (max gain around -16DB). I have the Cepstral Callie voice, and I must
> set the Cepstral volume to about 50% in order to match the gain of the
> English prompts. In conferences and other situations where prompts are
> played over conversations, the level of the prompts are obviously low.
> 
> I can, of course, renormalize the gain of the prompts up to -10DB or more
> with a sound editor. However, I wonder if there is a better way to change
> the level of the prompts, or if there is a good reason for the prompts to be
> encoded at such a low level.
> 
> I haven't considered all of the implications yet, but I'm fairly sure that
> encoding the prompts this quietly is not the best approach, even if it is
> desirable for the prompts to play quietly on a call. For each 6DB reduction
> in gain, there is a 50% reduction in perceived volume, and one less
> significant bit is used for storing the audio. In a 16-bit file, a maximum
> gain of -16 means that only the 14 least significant bits are actually used
> for encoding the audio. This results in a reduction in dynamic range, but
> the difference isn't really noticeable as long as the data remains 16-bit.
> The problem comes when the audio is converted to a different bit depth. For
> example, most quickie routines for converting 16-bit audio to 8-bit audio
> will simply chop off the 8 least significant bits. Therefore, when the
> prompts are converted to 8-bit audio for use by most of the narrow band
> codecs, the prompts are only using 6 bits of audio. If the volume of the
> channel is increased, then the 
> 6 bits are promoted, and the dithering errors at the bottom become louder.
> In the worst case, since these prompts are only encoded with 14 bits of
> actual data, and converting to an 8-bit channel will only leave 6 bits of
> actual data, boosting the gain of the prompts on an 8-bit channel to full
> loudness would result in the noise floor (the level of the dithering
> crackle) being about -18DB. That's almost as loud as the prompts themselves
> sound at the moment.
> 
> Anyway, regardless of observations, is there a reason why the prompts must
> be recorded so quietly? If I'd like them louder (without increasing the gain
> of the entire channel), is there a way other than running them through a
> tool to renormalize their max levels?
> 
> Bryan
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From bryansmart at bryansmart.com  Mon Jun 27 03:57:21 2011
From: bryansmart at bryansmart.com (Bryan Smart)
Date: Sun, 26 Jun 2011 19:57:21 -0400
Subject: [Freeswitch-users] Phrase macros in conferences
Message-ID: <FE02E1C7-64EA-4E85-AC33-B491AB2FF555@bryansmart.com>

I'm attempting to use phrase macros in mod_conference, but I'm not having success. Perhaps they aren't supported?

I have a working phrase macro. In the dialplan, both

<action application="phrase" data="my_phrase"/>

and

<action application="playback" data="phrase:my_phrase"/>

Work just fine.

The wiki says that, in conference.conf.xml, entries like pin-sound accept, for their value, file paths, strings prefixed with "say" for TTS, and strings prefixed with "tone_stream". This looks almost the same as what can be passed to the playback application in the dialplan. When I try to prefix a string with "phrase", it doesn't work.

<param name="pin-sound" value="phrase:my_phrase"/>

The console tells me "unknown file format". The console still tells me this if I make up phrase names

<param name="pin-sound" value="phrase:not_a_real_phrase"/>

So it is complaining about the syntax or capability, rather than an unrecognized format of an actual file.

If mod_conference doesn't support phrase macros, is this a near-future feature? Phrases would seem to be necessary to ocalize announcements and play announcements that are stitched together from multiple files.

Bryan

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From max.asterisk at gmail.com  Mon Jun 27 17:05:42 2011
From: max.asterisk at gmail.com (Max Alex)
Date: Mon, 27 Jun 2011 18:35:42 +0530
Subject: [Freeswitch-users] Answer issue on inbound call
In-Reply-To: <BANLkTinEp1Cu5mB_UHw-Y+mitGsNByR3sQ@mail.gmail.com>
References: <BANLkTin+f=uDMiOouyLWAorPvNTd1Z6xaA@mail.gmail.com>
	<BANLkTim=TOQD1tUvmHjeRuhAR-eKkGYgsw@mail.gmail.com>
	<BANLkTim3pTPV804t_Usk59eG2C56HAyFdA@mail.gmail.com>
	<BANLkTi=Q3qRBCDRARDWp15=Guwyq-jtNLQ@mail.gmail.com>
	<BANLkTinMokGeJU+wVVA0U-oxB97z+c+ZGg@mail.gmail.com>
	<BANLkTinEp1Cu5mB_UHw-Y+mitGsNByR3sQ@mail.gmail.com>
Message-ID: <BANLkTikcWDbCkTPRow-TMqPN=D3FV+LjWw@mail.gmail.com>

Hi,
Thanks for reply,
I have tried the same way and reloaded freeswitch, but still it is answered
on first ring of the call.
When it is ringing the call on 1001 and the same time it is answered on cell
phone, so something is done when it is ringing on 1001.

Here is the dialplan for the same

    <extension name="Local_Extension">
      <condition field="destination_number" expression="^(10[01][0-9])$">
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <action application="bind_meta_app" data="2 b s
record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
        <action application="bind_meta_app" data="3 b s
execute_extension::cf XML features"/>-->
        <action application="set" data="ringback=${us-ring}"/>
        <action application="set" data="transfer_ringback=$${hold_music}"/>
        <action application="set" data="call_timeout=30"/>
          <action application="set" data="hangup_after_bridge=true"/>
         <action application="set" data="continue_on_fail=true"/>
        <action application="hash"
data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
        <action application="hash"
data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
        <action application="set"
data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
var callgroup)}"/>
        <action application="hash"
data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
       <action application="bridge"
data="{ignore_early_media=true}user/${dialed_extension}@${domain_name}"/>
       <action application="sleep" data="1000"/>
        <action application="voicemail" data="default ${domain_name}
${dialed_extension}"/>
      </condition>
    </extension>


Please help me for this issue.

Thanks,
Max Alex
Voip Developer



On Fri, Jun 24, 2011 at 11:38 AM, Michael Collins <msc at freeswitch.org>wrote:

> Are you using the default dialplan? I think you might just need to ignore
> early media on your bridge app. If you are using the default.xml file then
> locate "Local_Extension" and the bridge line:
>
>     <action application="bridge" data="user/${dialed_extension}@
> ${domain_name}"/>
>
> Change it to this, then try again:
>
>     <action application="bridge"
> data="{ignore_early_media=true}user/${dialed_extension}@${domain_name}"/>
>
> If I understand correctly, the "symptom" you are experiencing is the normal
> operation of the bridge app (and it's cousin, the originate API command).
> When the b-leg sends back media, including ringing, the bridge (or the
> originate) will be considered "successful," and in the case of bridge, the
> b-leg's audio (the early media) will be connected to the a-leg. If you set
> ignore_early_media=true then you are explicitly telling the bridge app that
> you only want to connect the b-leg to the a-leg if the b-leg actually
> answers.
>
> I hope that made sense...
>
> -MC
>
>
>
> On Thu, Jun 23, 2011 at 9:32 PM, Max Alex <max.asterisk at gmail.com> wrote:
>
>> Hi,
>> Thanks for reply.
>> Current scenario is below.
>>
>> PSTN call -> sangoma device -> freeswitch incoming context -> routed to
>> 1001 -> ringing (Answered on cellphone)
>> Here when it is routed to 1001 the call it is started ringing, but on
>> phone that call is answered and hearding sound of ringing.
>>
>> Required flow:
>> PSTN call -> sangoma device -> freeswitch incoming context -> routed to
>> 1001 -> ringing (Ringing on cellphone)
>>
>> I hope it is understable, the call should not be answered until 1001
>> answer it, right not when it is started ring it is answered on cell phone.
>> that should not be happen as it is not answered yet.
>>
>> Waiting for your reply.
>>
>>
>> Thanks,
>> Max Alex
>> Voip Developer
>>
>>
>>
>> On Fri, Jun 24, 2011 at 5:48 AM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> I'm not sure I understand the problem. What is happening vs. what you
>>> believe should be happening?
>>> -MC
>>>
>>>
>>> On Thu, Jun 23, 2011 at 3:31 AM, Max Alex <max.asterisk at gmail.com>wrote:
>>>
>>>> Hi,
>>>> Thanks for your reply.
>>>> Here is my configuration and log
>>>> http://pastebin.freeswitch.org/16571
>>>>
>>>> I am using A200 analog sangoma device with freeswitch, it is working
>>>> fine but when it is routing call to 1001 then it is answered.
>>>> Please provider your suggestions.
>>>>
>>>> Thanks,
>>>> Max Alex
>>>> Voip Developer
>>>>
>>>>
>>>>
>>>>
>>>> On Wed, Jun 22, 2011 at 8:51 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>> I thought the A200 was an analog card? Maybe I have my numbers mixed
>>>>> up...
>>>>>
>>>>> Go ahead and collect a debug log of this call. It might help to have
>>>>> your configs posted as well. Use pastebin.freeswitch.org. See this
>>>>> wiki article for tips on how to collect information:
>>>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>>>>
>>>>> -MC
>>>>>
>>>>> On Wed, Jun 22, 2011 at 3:23 AM, Max Alex <max.asterisk at gmail.com>wrote:
>>>>>
>>>>>> Hi,
>>>>>> I have installed freeswitch latest version with sangoma card A200 as
>>>>>> well,
>>>>>> I have installed and configured freetdm module with wanpipe drivers
>>>>>> for freeswitch,
>>>>>> We are properly receiving the inbound calls in public context and then
>>>>>> we are routing that call to 1001 extension,
>>>>>> it is properly routing to 1001 as well, but we have one issue while
>>>>>> routing on 1001.
>>>>>>
>>>>>> Here is the issue description.
>>>>>> I am calling from my cell phone to that DID number of pri line, and
>>>>>> then it will start ringing on 1001 extension,
>>>>>> When 1001 extension start ringing the call is answered on my cell
>>>>>> phone,
>>>>>> it is something like freeswitch preanswer or autoanswer the call, how
>>>>>> can i stop this answer call when it is ringing on 1001 extension,
>>>>>> Waiting for good reply.
>>>>>>
>>>>>> Thanks,
>>>>>> Max Alex
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Mon Jun 27 17:25:04 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 27 Jun 2011 14:25:04 +0100
Subject: [Freeswitch-users] Reject SIP registrations
In-Reply-To: <BANLkTikU6fRYwmUTxzsSHyWs8raukLx4jw@mail.gmail.com>
References: <BANLkTikU6fRYwmUTxzsSHyWs8raukLx4jw@mail.gmail.com>
Message-ID: <BANLkTimq4YjhjX0vgCt3fsmBcAZRC1OZ-g@mail.gmail.com>

Just so you know...

     <param name="accept-blind-reg" value="true"/>
     <param name="accept-blind-auth" value="true"/>

These will have no effect in the user directory. They only apply to SIP
profiles.

-Steve



On 27 June 2011 02:23, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:

> Hello FS Users!
>
> I'm trying to create the following setup. When a user registers, if
> they register on a known account (lets say X), they do not need a
> password. X's registration is immediately OK'd, and everything is
> great. I've gotten that working using the ACL. The IP address of our
> SIP clients are added through cidr and the clients do not need to give
> passwords.
>
> However, for some reason, if another account that does not exist in
> the directory (let's say Y) registers, FS returns with a 200 OK,
> instead of rejecting Y. I'm trying to figure out why this is the case,
> and how to remedy that fact.
>
> I have the following line in my internal.xml file, which I had assumed
> would force this function:
>   <!-- Force the user and auth-user to match. -->
>   <param name="inbound-reg-force-matching-username" value="true"/>
>
> However, it does not work. In my directory, each individual account as
> the following lines:
>  <user id="1303">
>    <params>
>      <param name="accept-blind-reg" value="true"/>
>      <param name="accept-blind-auth" value="true"/>
>      <param name="vm-password" value="1000"/>
>    </params>
>
> Though I've found that removing it (from all users in the directory)
> doesn't help.
>
> I'm primarily concerned with the line in internal.xml; it seems
> possible that the fact that we do not have an auth-user (because we do
> not require auth) means that this won't work. However, I have yet to
> test that hypothesis. The ACL has been the most confusing aspect of
> this installation, with a lot of undocumented aspects, and I get the
> nagging feeling this is another. I could very well be wrong though.
>
> Thanks for any direction.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveu at coppice.org  Mon Jun 27 17:48:20 2011
From: steveu at coppice.org (Steve Underwood)
Date: Mon, 27 Jun 2011 21:48:20 +0800
Subject: [Freeswitch-users] Proper prompt gain/level
In-Reply-To: <66E81300-46E9-4C81-B2E0-123BCADDCAEB@bryansmart.com>
References: <FE081E27-8D62-4715-B7AA-64432598E013@bryansmart.com>	<0D3AEFF37C1643B79CE3F7A7EE9EAEC3@dell9400>
	<66E81300-46E9-4C81-B2E0-123BCADDCAEB@bryansmart.com>
Message-ID: <4E088A24.7010002@coppice.org>

On 06/27/2011 07:11 AM, Bryan Smart wrote:
> I have tools to batch-process audio files. I just was not sure that regaining all of the prompt files was the best approach. I figured that the gain must have been reduced so dramatically for some sort of reason (to avoid clipping in some situation, to work better with the internal resampling, etc).
>
> What AGC do you mean? I know that AGC has recently been added to conferencing, but the level of the prompts is a system-wide situation. As far as I know, there isn't AGC that can be applied on every channel, and, even if there was, there would surely be a processing hit, so the goal would be to avoid needing it, right?
>
> The root problem, at least for me, is this. I need to add voice prompts and other audio for an IVR. I can't simply normalize all of my prompts to 0DB, as, even though they don't distort, they're so loud when compared to the stock prompts, they'll blow the phone out of my hand. To match them to the stock prompts, I must normalize them to around -16DB. I can do that, but it seems very wrong. At -16DB, nearly 85% of the potential gain of the channel is lost.
-16dBM0 or -16dBOv, and average or peak burst power? -16dBOv for the 
average power is about where you want a voice prompt to be. In some 
juristictions you could be in breach of a regulation or two if you set 
the level higher than that on the PSTN. Why would you set a voice prompt 
to 0dB? It will be clipping like crazy.
> Try this... With the demo IVR (5000), add this before the sleep command in the dialplan.
>
> <action application="set_audio_level" data="write 4"/>
>
> That is the max gain boost available for a channel. The prompts should be really clipping with that much amplification, but they don't clip at all. At -16DB, you could literally amplify them to 6 times their native level without distorting. Native level is too low. Once I realized this, it became clear to me why Freeswitch sounded more quiet than Asterisk, at least when working with recorded prompts.
>
> I suppose I could use set_audio_level on every last call, but I'm sure that real-time amplification, like AGC, is another processor drain that builds up with lots of calls. Besides, it seems weird to dramatically reduce the level of audio, and then waste cycles amplifying it back up in real-time.
>
> Bryan
Steve



From msc at freeswitch.org  Mon Jun 27 19:09:10 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 27 Jun 2011 08:09:10 -0700
Subject: [Freeswitch-users] Leg-A is automatically disconnected on Leg-B
 orginate failure
In-Reply-To: <BANLkTinZJf3BxB7QHefnmOvxQmVW2r4dTg@mail.gmail.com>
References: <1308213151543-6482192.post@n2.nabble.com>
	<BANLkTi=xYgBTPVGxVGmk9f2=bmX5X2vF=Q@mail.gmail.com>
	<BANLkTikhPmjN5HeeJHBnQt1ivA8zbtCnYA@mail.gmail.com>
	<BANLkTi=m7oMix1ZRC9URtRuAbsG2fWyhMQ@mail.gmail.com>
	<BANLkTimxBU8m311LVKMWcFOjukwLg+ZUGQ@mail.gmail.com>
	<BANLkTinZJf3BxB7QHefnmOvxQmVW2r4dTg@mail.gmail.com>
Message-ID: <BANLkTimiExLew_1cjMuQVpF3CKkR8mrGXQ@mail.gmail.com>

This makes my eyes bleed. Can you please put this on pastebin.freeswitch.org?
Use "FreeSWITCH Log" as the syntax highlight.
-MC

On Sun, Jun 26, 2011 at 11:27 PM, David Ma <mays.david at gmail.com> wrote:

> Hi Michael,
>
> Thanks very much for the response. There is no dialplan specified for this
> call. In the FS setting, G729 is used as preferred stack for originating
> calls. The leg-A and B are bridged immediately after receiving
> PROGRESS-MEDIA from leg-B.
>
> Calls are originated with following parameters:
> *
> -- Leg-A --
>             "api originate
> {origination_caller_id_number=<caller>,sip_cid_type=pid,privacy=yes,continue_on_fail=true}
> <called> &park()"
>
> **-- Leg-B --*
> *            "api originate
> {origination_caller_id_number=<caller>,originate_timeout=60,sip_cid_type=pid,privacy=yes,continue_on_fail=false}
> <called> &park()"
> *
>
> The entire debug log for this call follows.
>
> Thanks,
> D.Ma
>
> =================================
> 2011-06-24 13:29:35.887830 [DEBUG] switch_ivr_originate.c:1971 variable
> string 0 = [origination_caller_id_number=03996563750911]
> 2011-06-24 13:29:35.887830 [DEBUG] switch_ivr_originate.c:1971 variable
> string 1 = [originate_timeout=60]
> 2011-06-24 13:29:35.887830 [DEBUG] switch_ivr_originate.c:1971 variable
> string 2 = [ccd_session_id=20110624132936888918]
> 2011-06-24 13:29:35.887830 [DEBUG] switch_ivr_originate.c:1971 variable
> string 3 = [sip_cid_type=pid]
> 2011-06-24 13:29:35.887830 [DEBUG] switch_ivr_originate.c:1971 variable
> string 4 = [privacy=yes]
> 2011-06-24 13:29:35.887830 [DEBUG] switch_ivr_originate.c:1971 variable
> string 5 = [continue_on_fail=true]
> 2011-06-24 13:29:35.887830 [NOTICE] switch_channel.c:808 New Channel
> sofia/external/03996597632298 at 203.208.207.212[c0bd700d-913c-42ad-b68f-81001bf658b8]
> 2011-06-24 13:29:35.887830 [DEBUG] mod_sofia.c:4129 (sofia/external/
> 03996597632298 at 203.208.207.212) State Change CS_NEW -> CS_INIT
> 2011-06-24 13:29:35.887830 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996597632298 at 203.208.207.212 [BREAK]
> 2011-06-24 13:29:35.889072 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996597632298 at 203.208.207.212) Running State Change
> CS_INIT
> 2011-06-24 13:29:35.889829 [DEBUG] switch_core_state_machine.c:356
> (sofia/external/03996597632298 at 203.208.207.212) State INIT
> 2011-06-24 13:29:35.889829 [DEBUG] mod_sofia.c:84 sofia/external/
> 03996597632298 at 203.208.207.212 SOFIA INIT
> send 999 bytes to udp/[203.208.207.212]:5060 at 05:29:36.398490:
>    ------------------------------------------------------------------------
>    INVITE sip:03996597632298 at 203.208.207.212 SIP/2.0
>    Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKtpaQ28SQXKvem
>    Max-Forwards: 70
>    From: "" <sip:03996563750911 at 202.73.56.46>;tag=5SNjUy956H2mm
>    To: <sip:03996597632298 at 203.208.207.212>
>    Call-ID: ca5d29b7-18c5-122f-3387-0015c5fc7ad9
>    CSeq: 14110624 INVITE
>    Contact: <sip:mod_sofia at 202.73.56.46:5080>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, PRACK, NOTIFY
>    Supported: 100rel, timer, precondition, path, replaces
>    Allow-Events: talk, hold, refer
>    Privacy: none
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 204
>    X-FS-Support: update_display
>    P-Asserted-Identity: "" <sip:03996563750911 at 202.73.56.46>
>
>    v=0
>    o=FreeSWITCH 1308862405 1308862406 IN IP4 202.73.56.46
>    s=FreeSWITCH
>    c=IN IP4 202.73.56.46
>    t=0 0
>    m=audio 30970 RTP/AVP 18 3 101 13
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
>    ------------------------------------------------------------------------
> 2011-06-24 13:29:35.891343 [DEBUG] mod_sofia.c:124 (sofia/external/
> 03996597632298 at 203.208.207.212) State Change CS_INIT -> CS_ROUTING
> 2011-06-24 13:29:35.891343 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996597632298 at 203.208.207.212 [BREAK]
> 2011-06-24 13:29:35.891343 [DEBUG] sofia.c:4646 Channel sofia/external/
> 03996597632298 at 203.208.207.212 entering state [calling][0]
> 2011-06-24 13:29:35.891343 [DEBUG] switch_core_state_machine.c:356
> (sofia/external/03996597632298 at 203.208.207.212) State INIT going to sleep
> 2011-06-24 13:29:35.891343 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996597632298 at 203.208.207.212) Running State Change
> CS_ROUTING
> 2011-06-24 13:29:35.891343 [DEBUG] switch_channel.c:1657 (sofia/external/
> 03996597632298 at 203.208.207.212) Callstate Change DOWN -> RINGING
> 2011-06-24 13:29:35.891343 [DEBUG] switch_core_state_machine.c:359
> (sofia/external/03996597632298 at 203.208.207.212) State ROUTING
> 2011-06-24 13:29:35.891343 [DEBUG] mod_sofia.c:147 sofia/external/
> 03996597632298 at 203.208.207.212 SOFIA ROUTING
> 2011-06-24 13:29:35.891343 [DEBUG] switch_ivr_originate.c:66
> (sofia/external/03996597632298 at 203.208.207.212) State Change CS_ROUTING ->
> CS_CONSUME_MEDIA
> 2011-06-24 13:29:35.891343 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996597632298 at 203.208.207.212 [BREAK]
> 2011-06-24 13:29:35.891343 [DEBUG] switch_core_state_machine.c:359
> (sofia/external/03996597632298 at 203.208.207.212) State ROUTING going to
> sleep
> 2011-06-24 13:29:35.891343 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996597632298 at 203.208.207.212) Running State Change
> CS_CONSUME_MEDIA
> 2011-06-24 13:29:35.891343 [DEBUG] switch_core_state_machine.c:378
> (sofia/external/03996597632298 at 203.208.207.212) State CONSUME_MEDIA
> 2011-06-24 13:29:35.891343 [DEBUG] switch_core_state_machine.c:378
> (sofia/external/03996597632298 at 203.208.207.212) State CONSUME_MEDIA going
> to sleep
> recv 307 bytes from udp/[203.208.207.212]:5060 at 05:29:36.404598:
>    ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/UDP 10.1.1.46:5080
> ;rport=5080;branch=z9hG4bKtpaQ28SQXKvem;received=10.1.1.46
>    From: "" <sip:03996563750911 at 10.1.1.46>;tag=5SNjUy956H2mm
>    To: <sip:03996597632298 at 203.208.207.212>
>    Call-ID: ca5d29b7-18c5-122f-3387-0015c5fc7ad9
>    CSeq: 14110624 INVITE
>    Content-Length:     0
>
>    ------------------------------------------------------------------------
> recv 774 bytes from udp/[203.208.207.212]:5060 at 05:29:37.615248:
>    ------------------------------------------------------------------------
>    SIP/2.0 183 Session Progress
>    Via: SIP/2.0/UDP 10.1.1.46:5080
> ;rport=5080;branch=z9hG4bKtpaQ28SQXKvem;received=10.1.1.46
>    From: "" <sip:03996563750911 at 10.1.1.46>;tag=5SNjUy956H2mm
>    To: <sip:03996597632298 at 203.208.207.212
> >;tag=2QGB951HCR30000E1D0001Ll000B8RX0S6R1TX
>    Call-ID: ca5d29b7-18c5-122f-3387-0015c5fc7ad9
>    CSeq: 14110624 INVITE
>    Contact: <sip:203.208.207.212:5060>
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
> SUBSCRIBE, UPDATE
>    Content-Type: application/sdp
>    Require: 100rel
>    RSeq: 1
>    Content-Length:   231
>
>    v=0
>    o=- 508539113671071081 1 IN IP4 203.208.207.212
>    s=session
>    c=IN IP4 203.208.207.201
>    t=0 0
>    m=audio 25120 RTP/AVP 18 101
>    a=rtpmap:18 G729/8000
>    a=fmtp:18 annexb=no
>    a=rtpmap:101 telephone-event/8000
>    a=ptime:20
>    a=sendrecv
>    ------------------------------------------------------------------------
> send 651 bytes to udp/[203.208.207.212]:5060 at 05:29:37.615605:
>    ------------------------------------------------------------------------
>    PRACK sip:203.208.207.212:5060 SIP/2.0
>    Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKUZ3F43aUtvj1F
>    Max-Forwards: 70
>    From: "" <sip:03996563750911 at 202.73.56.46>;tag=5SNjUy956H2mm
>    To: <sip:03996597632298 at 203.208.207.212
> >;tag=2QGB951HCR30000E1D0001Ll000B8RX0S6R1TX
>    Call-ID: ca5d29b7-18c5-122f-3387-0015c5fc7ad9
>    CSeq: 14110625 PRACK
>    Contact: <sip:mod_sofia at 202.73.56.46:5080>
>    RAck: 1 14110624 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, PRACK, NOTIFY
>    Supported: 100rel, timer, precondition, path, replaces
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> 2011-06-24 13:29:37.107576 [INFO] sofia.c:729 sofia/external/
> 03996597632298 at 203.208.207.212 Update Callee ID to "Outbound Call"
> <03996597632298>
> 2011-06-24 13:29:37.107576 [DEBUG] sofia.c:4646 Channel sofia/external/
> 03996597632298 at 203.208.207.212 entering state [proceeding][183]
> 2011-06-24 13:29:37.107576 [DEBUG] sofia.c:4657 Remote SDP:
> v=0
> o=- 508539113671071081 1 IN IP4 203.208.207.212
> s=session
> c=IN IP4 203.208.207.201
> t=0 0
> m=audio 25120 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=ptime:20
>
> 2011-06-24 13:29:37.107576 [DEBUG] sofia_glue.c:4467 Audio Codec Compare
> [G729:18:8000:20:8000]/[G729:18:8000:20:8000]
> 2011-06-24 13:29:37.107576 [DEBUG] sofia_glue.c:2757 Set Codec
> sofia/external/03996597632298 at 203.208.207.212 G729/8000 20 ms 160 samples
> 8000 bits
> 2011-06-24 13:29:37.108995 [DEBUG] sofia_glue.c:4565 Set 2833 dtmf send
> payload to 101
> 2011-06-24 13:29:37.108995 [DEBUG] sofia_glue.c:2987 AUDIO RTP
> [sofia/external/03996597632298 at 203.208.207.212] 10.1.1.46 port 30970 ->
> 203.208.207.201 port 25120 codec: 18 ms: 20
> 2011-06-24 13:29:37.108995 [DEBUG] switch_rtp.c:1607 Starting timer [soft]
> 160 bytes per 20ms
> 2011-06-24 13:29:37.109893 [DEBUG] sofia_glue.c:3228 Set 2833 dtmf send
> payload to 101
> 2011-06-24 13:29:37.109893 [DEBUG] sofia_glue.c:3233 Set 2833 dtmf receive
> payload to 101
> 2011-06-24 13:29:37.109893 [NOTICE] sofia_glue.c:3680 Pre-Answer
> sofia/external/03996597632298 at 203.208.207.212!
> 2011-06-24 13:29:37.109893 [DEBUG] switch_channel.c:2627 (sofia/external/
> 03996597632298 at 203.208.207.212) Callstate Change RINGING -> EARLY
> 2011-06-24 13:29:37.109893 [DEBUG] switch_ivr_originate.c:3408 Originate
> Resulted in Success: [sofia/external/03996597632298 at 203.208.207.212]
> 2011-06-24 13:29:37.109893 [DEBUG] mod_commands.c:3205 (sofia/external/
> 03996597632298 at 203.208.207.212) State Change CS_CONSUME_MEDIA ->
> CS_EXECUTE
> 2011-06-24 13:29:37.109893 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996597632298 at 203.208.207.212 [BREAK]
> 2011-06-24 13:29:37.111618 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996597632298 at 203.208.207.212) Running State Change
> CS_EXECUTE
> 2011-06-24 13:29:37.111618 [DEBUG] switch_core_state_machine.c:366
> (sofia/external/03996597632298 at 203.208.207.212) State EXECUTE
> 2011-06-24 13:29:37.111618 [DEBUG] mod_sofia.c:240 sofia/external/
> 03996597632298 at 203.208.207.212 SOFIA EXECUTE
> 2011-06-24 13:29:37.111618 [DEBUG] switch_core_state_machine.c:157
> sofia/external/03996597632298 at 203.208.207.212 Standard EXECUTE
> EXECUTE sofia/external/03996597632298 at 203.208.207.212 park()
> recv 382 bytes from udp/[203.208.207.212]:5060 at 05:29:37.620736:
>    ------------------------------------------------------------------------
>    SIP/2.0 200 Ok
>    Via: SIP/2.0/UDP 10.1.1.46:5080
> ;rport=5080;branch=z9hG4bKUZ3F43aUtvj1F;received=10.1.1.46
>    From: "" <sip:03996563750911 at 10.1.1.46>;tag=5SNjUy956H2mm
>    To: <sip:03996597632298 at 203.208.207.212
> >;tag=2QGB951HCR30000E1D0001Ll000B8RX0S6R1TX
>    Call-ID: ca5d29b7-18c5-122f-3387-0015c5fc7ad9
>    CSeq: 14110625 PRACK
>    Contact: <sip:203.208.207.212:5060>
>    Content-Length:     0
>
>    ------------------------------------------------------------------------
> 2011-06-24 13:29:37.160714 [DEBUG] switch_rtp.c:2933 Correct ip/port
> confirmed.
> recv 512 bytes from udp/[203.208.207.212]:5060 at 05:29:45.146388:
>    ------------------------------------------------------------------------
>    SIP/2.0 183 Session Progress
>    Via: SIP/2.0/UDP 10.1.1.46:5080
> ;rport=5080;branch=z9hG4bKtpaQ28SQXKvem;received=10.1.1.46
>    From: "" <sip:03996563750911 at 10.1.1.46>;tag=5SNjUy956H2mm
>    To: <sip:03996597632298 at 203.208.207.212
> >;tag=2QGB951HCR30000E1D0001Ll000B8RX0S6R1TX
>    Call-ID: ca5d29b7-18c5-122f-3387-0015c5fc7ad9
>    CSeq: 14110624 INVITE
>    Contact: <sip:203.208.207.212:5060>
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
> SUBSCRIBE, UPDATE
>    Require: 100rel
>    RSeq: 2
>    Content-Length:     0
>
>    ------------------------------------------------------------------------
> send 651 bytes to udp/[203.208.207.212]:5060 at 05:29:45.146628:
>    ------------------------------------------------------------------------
>    PRACK sip:203.208.207.212:5060 SIP/2.0
>    Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKv8v85yUyQ58KB
>   Max-Forwards: 70
>    From: "" <sip:03996563750911 at 202.73.56.46>;tag=5SNjUy956H2mm
>    To: <sip:03996597632298 at 203.208.207.212
> >;tag=2QGB951HCR30000E1D0001Ll000B8RX0S6R1TX
>    Call-ID: ca5d29b7-18c5-122f-3387-0015c5fc7ad9
>    CSeq: 14110626 PRACK
>    Contact: <sip:mod_sofia at 202.73.56.46:5080>
>    RAck: 2 14110624 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, PRACK, NOTIFY
>    Supported: 100rel, timer, precondition, path, replaces
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> 2011-06-24 13:29:44.639310 [INFO] sofia.c:729 sofia/external/
> 03996597632298 at 203.208.207.212 Update Callee ID to "03996597632298"
> <03996597632298>
> 2011-06-24 13:29:44.639310 [DEBUG] sofia.c:4641 Channel sofia/external/
> 03996597632298 at 203.208.207.212 skipping state [proceeding][183]
> recv 382 bytes from udp/[203.208.207.212]:5060 at 05:29:45.151718:
>    ------------------------------------------------------------------------
>    SIP/2.0 200 Ok
>    Via: SIP/2.0/UDP 10.1.1.46:5080
> ;rport=5080;branch=z9hG4bKv8v85yUyQ58KB;received=10.1.1.46
>    From: "" <sip:03996563750911 at 10.1.1.46>;tag=5SNjUy956H2mm
>    To: <sip:03996597632298 at 203.208.207.212
> >;tag=2QGB951HCR30000E1D0001Ll000B8RX0S6R1TX
>    Call-ID: ca5d29b7-18c5-122f-3387-0015c5fc7ad9
>    CSeq: 14110626 PRACK
>    Contact: <sip:203.208.207.212:5060>
>    Content-Length:     0
>
>    ------------------------------------------------------------------------
> recv 529 bytes from udp/[203.208.207.212]:5060 at 05:29:47.446072:
>    ------------------------------------------------------------------------
>    SIP/2.0 200 Ok
>    Via: SIP/2.0/UDP 10.1.1.46:5080
> ;rport=5080;branch=z9hG4bKtpaQ28SQXKvem;received=10.1.1.46
>    From: "" <sip:03996563750911 at 10.1.1.46>;tag=5SNjUy956H2mm
>    To: <sip:03996597632298 at 203.208.207.212
> >;tag=2QGB951HCR30000E1D0001Ll000B8RX0S6R1TX
>    Call-ID: ca5d29b7-18c5-122f-3387-0015c5fc7ad9
>    CSeq: 14110624 INVITE
>    Contact: <sip:203.208.207.212:5060>
>    Allow-Events: refer
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
> SUBSCRIBE, UPDATE
>    Supported: 100rel, timer, replaces
>    Content-Length:     0
>
>    ------------------------------------------------------------------------
> 2011-06-24 13:29:46.938477 [DEBUG] sofia.c:4646 Channel sofia/external/
> 03996597632298 at 203.208.207.212 entering state [completing][200]
> send 405 bytes to udp/[203.208.207.212]:5060 at 05:29:47.446869:
>    ------------------------------------------------------------------------
>    ACK sip:203.208.207.212:5060 SIP/2.0
>    Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKXHp17Sc2meZ6p
>    Max-Forwards: 70
>    From: "" <sip:03996563750911 at 202.73.56.46>;tag=5SNjUy956H2mm
>    To: <sip:03996597632298 at 203.208.207.212
> >;tag=2QGB951HCR30000E1D0001Ll000B8RX0S6R1TX
>    Call-ID: ca5d29b7-18c5-122f-3387-0015c5fc7ad9
>    CSeq: 14110624 ACK
>    Contact: <sip:mod_sofia at 202.73.56.46:5080>
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> 2011-06-24 13:29:46.939738 [DEBUG] sofia.c:4646 Channel sofia/external/
> 03996597632298 at 203.208.207.212 entering state [ready][200]
> 2011-06-24 13:29:46.939738 [DEBUG] switch_channel.c:2782 (sofia/external/
> 03996597632298 at 203.208.207.212) Callstate Change EARLY -> ACTIVE
> 2011-06-24 13:29:46.939738 [NOTICE] sofia.c:5175 Channel [sofia/external/
> 03996597632298 at 203.208.207.212] has been answered
> 2011-06-24 13:29:46.942763 [DEBUG] switch_scheduler.c:214 Added task 27
> switch_ivr_schedule_hangup (c0bd700d-913c-42ad-b68f-81001bf658b8) to run at
> 1308896986
> 2011-06-24 13:29:46.944563 [DEBUG] switch_core_session.c:954 Send signal
> sofia/external/03996597632298 at 203.208.207.212 [BREAK]
> 2011-06-24 13:29:46.945633 [DEBUG] switch_ivr_originate.c:1971 variable
> string 0 = [origination_caller_id_number=03996597632298]
> 2011-06-24 13:29:46.945633 [DEBUG] switch_ivr_originate.c:1971 variable
> string 1 = [ccd_session_id=20110624132936888918]
> 2011-06-24 13:29:46.945633 [DEBUG] switch_ivr_originate.c:1971 variable
> string 2 = [sip_cid_type=pid]
> 2011-06-24 13:29:46.945633 [DEBUG] switch_ivr_originate.c:1971 variable
> string 3 = [privacy=yes]
> 2011-06-24 13:29:46.945633 [DEBUG] switch_ivr_originate.c:1971 variable
> string 4 = [continue_on_fail=false]
> 2011-06-24 13:29:46.945633 [NOTICE] switch_channel.c:808 New Channel
> sofia/external/03996563750911 at 203.208.207.212[817124e6-26fe-46cc-af55-89715abdfced]
> 2011-06-24 13:29:46.945633 [DEBUG] mod_sofia.c:4129 (sofia/external/
> 03996563750911 at 203.208.207.212) State Change CS_NEW -> CS_INIT
> 2011-06-24 13:29:46.945633 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996563750911 at 203.208.207.212 [BREAK]
> 2011-06-24 13:29:46.945633 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996563750911 at 203.208.207.212) Running State Change
> CS_INIT
> 2011-06-24 13:29:46.945633 [DEBUG] switch_core_state_machine.c:356
> (sofia/external/03996563750911 at 203.208.207.212) State INIT
> 2011-06-24 13:29:46.945633 [DEBUG] mod_sofia.c:84 sofia/external/
> 03996563750911 at 203.208.207.212 SOFIA INIT
> 2011-06-24 13:29:46.945633 [DEBUG] mod_sofia.c:124 (sofia/external/
> 03996563750911 at 203.208.207.212) State Change CS_INIT -> CS_ROUTING
> 2011-06-24 13:29:46.945633 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996563750911 at 203.208.207.212 [BREAK]
> 2011-06-24 13:29:46.945633 [DEBUG] switch_core_state_machine.c:356
> (sofia/external/03996563750911 at 203.208.207.212) State INIT going to sleep
> 2011-06-24 13:29:46.945633 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996563750911 at 203.208.207.212) Running State Change
> CS_ROUTING
> 2011-06-24 13:29:46.945633 [DEBUG] switch_channel.c:1657 (sofia/external/
> 03996563750911 at 203.208.207.212) Callstate Change DOWN -> RINGING
> 2011-06-24 13:29:46.945633 [DEBUG] switch_core_state_machine.c:359
> (sofia/external/03996563750911 at 203.208.207.212) State ROUTING
> 2011-06-24 13:29:46.945633 [DEBUG] mod_sofia.c:147 sofia/external/
> 03996563750911 at 203.208.207.212 SOFIA ROUTING
> 2011-06-24 13:29:46.945633 [DEBUG] switch_ivr_originate.c:66
> (sofia/external/03996563750911 at 203.208.207.212) State Change CS_ROUTING ->
> CS_CONSUME_MEDIA
> 2011-06-24 13:29:46.945633 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996563750911 at 203.208.207.212 [BREAK]
> 2011-06-24 13:29:46.945633 [DEBUG] switch_core_state_machine.c:359
> (sofia/external/03996563750911 at 203.208.207.212) State ROUTING going to
> sleep
> 2011-06-24 13:29:46.945633 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996563750911 at 203.208.207.212) Running State Change
> CS_CONSUME_MEDIA
> send 999 bytes to udp/[203.208.207.212]:5060 at 05:29:47.454251:
>    ------------------------------------------------------------------------
>    INVITE sip:03996563750911 at 203.208.207.212 SIP/2.0
>    Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKytFt9mX5HQNSj
>    Max-Forwards: 70
>    From: "" <sip:03996597632298 at 202.73.56.46>;tag=62eBXSt93tr7F
>    To: <sip:03996563750911 at 203.208.207.212>
>    Call-ID: d0f42376-18c5-122f-3387-0015c5fc7ad92011-06-24 13:29:46.945633
> [DEBUG] switch_core_state_machine.c:378 (sofia/external/
> 03996563750911 at 203.208.207.212) State CONSUME_MEDIA
>
>    CSeq: 14110629 INVITE
>    Contact: <sip:mod_sofia at 202.73.56.46:5080>
> 2011-06-24 13:29:46.945633 [DEBUG] switch_core_state_machine.c:378
> (sofia/external/03996563750911 at 203.208.207.212) State CONSUME_MEDIA going
> to sleep
>    User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, PRACK, NOTIFY
>    Supported: 100rel, timer, precondition, path, replaces
>    Allow-Events: talk, hold, refer
>    Privacy: none
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 204
>    X-FS-Support: update_display
>    P-Asserted-Identity: "" <sip:03996597632298 at 202.73.56.46>
>
>    v=0
>    o=FreeSWITCH 1308872756 1308872757 IN IP4 202.73.56.46
>    s=FreeSWITCH
>    c=IN IP4 202.73.56.46
>    t=0 0
>    m=audio 20630 RTP/AVP 18 3 101 13
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
>    ------------------------------------------------------------------------
> 2011-06-24 13:29:46.946974 [DEBUG] sofia.c:4646 Channel sofia/external/
> 03996563750911 at 203.208.207.212 entering state [calling][0]
> recv 307 bytes from udp/[203.208.207.212]:5060 at 05:29:47.460017:
>    ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/UDP 10.1.1.46:5080
> ;rport=5080;branch=z9hG4bKytFt9mX5HQNSj;received=10.1.1.46
>    From: "" <sip:03996597632298 at 10.1.1.46>;tag=62eBXSt93tr7F
>    To: <sip:03996563750911 at 203.208.207.212>
>    Call-ID: d0f42376-18c5-122f-3387-0015c5fc7ad9
>    CSeq: 14110629 INVITE
>    Content-Length:     0
>
>    ------------------------------------------------------------------------
> 2011-06-24 13:29:46.960900 [DEBUG] switch_ivr.c:563 sofia/external/
> 03996597632298 at 203.208.207.212 Command Execute
> playback(/usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav)
> EXECUTE sofia/external/03996597632298 at 203.208.207.212playback(/usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav)
> 2011-06-24 13:29:46.960900 [DEBUG] switch_core_file.c:176 File
> /usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav sample rate 11025
> doesn't match requested rate 8000
> 2011-06-24 13:29:46.960900 [WARNING] switch_core_file.c:189 File has 2
> channels, muxing to mono will occur.
> 2011-06-24 13:29:46.960900 [DEBUG] switch_ivr_play_say.c:1244 Codec
> Activated L16 at 8000hz 2 channels 20ms
> 2011-06-24 13:29:46.981203 [INFO] mod_com_g729.c:119 ENCODER CREATE -
> 0x2aaab80894b8 0x5112110
> recv 774 bytes from udp/[203.208.207.212]:5060 at 05:29:48.623941:
>    ------------------------------------------------------------------------
>    SIP/2.0 183 Session Progress
>    Via: SIP/2.0/UDP 10.1.1.46:5080
> ;rport=5080;branch=z9hG4bKytFt9mX5HQNSj;received=10.1.1.46
>    From: "" <sip:03996597632298 at 10.1.1.46>;tag=62eBXSt93tr7F
>    To: <sip:03996563750911 at 203.208.207.212
> >;tag=2QGB951HCR30000E1D0001Ll000B8RV0KORLQK
>    Call-ID: d0f42376-18c5-122f-3387-0015c5fc7ad9
>    CSeq: 14110629 INVITE
>    Contact: <sip:203.208.207.212:5060>
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
> SUBSCRIBE, UPDATE
>    Content-Type: application/sdp
>    Require: 100rel
>    RSeq: 1
>    Content-Length:   231
>
>    v=0
>    o=- 508550174225779858 1 IN IP4 203.208.207.212
>    s=session
>    c=IN IP4 203.208.207.202
>    t=0 0
>    m=audio 25150 RTP/AVP 18 101
>    a=rtpmap:18 G729/8000
>    a=fmtp:18 annexb=no
>    a=rtpmap:101 telephone-event/8000
>    a=ptime:20
>    a=sendrecv
>    ------------------------------------------------------------------------
> send 651 bytes to udp/[203.208.207.212]:5060 at 05:29:48.624297:
>    ------------------------------------------------------------------------
>    PRACK sip:203.208.207.212:5060 SIP/2.0
>    Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKZ38jBge9e0Bce
>    Max-Forwards: 70
>    From: "" <sip:03996597632298 at 202.73.56.46>;tag=62eBXSt93tr7F
>    To: <sip:03996563750911 at 203.208.207.212
> >;tag=2QGB951HCR30000E1D0001Ll000B8RV0KORLQK
>    Call-ID: d0f42376-18c5-122f-3387-0015c5fc7ad9
>    CSeq: 14110630 PRACK
>    Contact: <sip:mod_sofia at 202.73.56.46:5080>
>    RAck: 1 14110629 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, PRACK, NOTIFY
>    Supported: 100rel, timer, precondition, path, replaces
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> 2011-06-24 13:29:48.115941 [INFO] sofia.c:729 sofia/external/
> 03996563750911 at 203.208.207.212 Update Callee ID to "Outbound Call"
> <03996563750911>
> 2011-06-24 13:29:48.115941 [DEBUG] sofia.c:4646 Channel sofia/external/
> 03996563750911 at 203.208.207.212 entering state [proceeding][183]
> 2011-06-24 13:29:48.115941 [DEBUG] sofia.c:4657 Remote SDP:
> v=0
> o=- 508550174225779858 1 IN IP4 203.208.207.212
> s=session
> c=IN IP4 203.208.207.202
> t=0 0
> m=audio 25150 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=ptime:20
>
> 2011-06-24 13:29:48.115941 [DEBUG] sofia_glue.c:4467 Audio Codec Compare
> [G729:18:8000:20:8000]/[G729:18:8000:20:8000]
> 2011-06-24 13:29:48.115941 [DEBUG] sofia_glue.c:2757 Set Codec
> sofia/external/03996563750911 at 203.208.207.212 G729/8000 20 ms 160 samples
> 8000 bits
> 2011-06-24 13:29:48.115941 [DEBUG] sofia_glue.c:4565 Set 2833 dtmf send
> payload to 101
> 2011-06-24 13:29:48.115941 [DEBUG] sofia_glue.c:2987 AUDIO RTP
> [sofia/external/03996563750911 at 203.208.207.212] 10.1.1.46 port 20630 ->
> 203.208.207.202 port 25150 codec: 18 ms: 20
> 2011-06-24 13:29:48.115941 [DEBUG] switch_rtp.c:1607 Starting timer [soft]
> 160 bytes per 20ms
> 2011-06-24 13:29:48.119065 [DEBUG] sofia_glue.c:3228 Set 2833 dtmf send
> payload to 101
> 2011-06-24 13:29:48.119065 [DEBUG] sofia_glue.c:3233 Set 2833 dtmf receive
> payload to 101
> 2011-06-24 13:29:48.119065 [NOTICE] sofia_glue.c:3680 Pre-Answer
> sofia/external/03996563750911 at 203.208.207.212!
> 2011-06-24 13:29:48.119065 [DEBUG] switch_channel.c:2627 (sofia/external/
> 03996563750911 at 203.208.207.212) Callstate Change RINGING -> EARLY
> 2011-06-24 13:29:48.120495 [DEBUG] switch_ivr_originate.c:3408 Originate
> Resulted in Success: [sofia/external/03996563750911 at 203.208.207.212]
> 2011-06-24 13:29:48.120495 [DEBUG] mod_commands.c:3205 (sofia/external/
> 03996563750911 at 203.208.207.212) State Change CS_CONSUME_MEDIA ->
> CS_EXECUTE
> 2011-06-24 13:29:48.120495 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996563750911 at 203.208.207.212 [BREAK]
> 2011-06-24 13:29:48.120495 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996563750911 at 203.208.207.212) Running State Change
> CS_EXECUTE
> 2011-06-24 13:29:48.120495 [DEBUG] switch_core_state_machine.c:366
> (sofia/external/03996563750911 at 203.208.207.212) State EXECUTE
> 2011-06-24 13:29:48.120495 [DEBUG] mod_sofia.c:240 sofia/external/
> 03996563750911 at 203.208.207.212 SOFIA EXECUTE
> 2011-06-24 13:29:48.120495 [DEBUG] switch_core_state_machine.c:157
> sofia/external/03996563750911 at 203.208.207.212 Standard EXECUTE
> EXECUTE sofia/external/03996563750911 at 203.208.207.212 park()
> 2011-06-24 13:29:48.122782 [DEBUG] switch_core_session.c:954 Send signal
> sofia/external/03996597632298 at 203.208.207.212 [BREAK]
> recv 382 bytes from udp/[203.208.207.212]:5060 at 05:29:48.630447:
>    ------------------------------------------------------------------------
>    SIP/2.0 200 Ok
>    Via: SIP/2.0/UDP 10.1.1.46:5080
> ;rport=5080;branch=z9hG4bKZ38jBge9e0Bce;received=10.1.1.46
>    From: "" <sip:03996597632298 at 10.1.1.46>;tag=62eBXSt93tr7F
>    To: <sip:03996563750911 at 203.208.207.212
> >;tag=2QGB951HCR30000E1D0001Ll000B8RV0KORLQK
>    Call-ID: d0f42376-18c5-122f-3387-0015c5fc7ad9
>    CSeq: 14110630 PRACK
>    Contact: <sip:203.208.207.212:5060>
>    Content-Length:     0
>
>    ------------------------------------------------------------------------
> 2011-06-24 13:29:48.122782 [DEBUG] switch_ivr.c:563 sofia/external/
> 03996597632298 at 203.208.207.212 Command Execute
> playback(tone_stream://%(2000,4000,440,480);loops=10)
> EXECUTE sofia/external/03996597632298 at 203.208.207.212playback(tone_stream://%(2000,4000,440,480);loops=10)
> 2011-06-24 13:29:48.124434 [DEBUG] switch_ivr_bridge.c:1480
> (sofia/external/03996597632298 at 203.208.207.212) State Change CS_EXECUTE ->
> CS_HIBERNATE
> 2011-06-24 13:29:48.124434 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996597632298 at 203.208.207.212 [BREAK]
> 2011-06-24 13:29:48.124434 [DEBUG] switch_ivr_bridge.c:1482
> (sofia/external/03996563750911 at 203.208.207.212) State Change CS_EXECUTE ->
> CS_HIBERNATE
> 2011-06-24 13:29:48.124434 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996563750911 at 203.208.207.212 [BREAK]
> 2011-06-24 13:29:48.124434 [DEBUG] switch_core_session.c:771 Send signal
> sofia/external/03996563750911 at 203.208.207.212 [BREAK]
> 2011-06-24 13:29:48.124434 [DEBUG] switch_core_session.c:771 Send signal
> sofia/external/03996597632298 at 203.208.207.212 [BREAK]
> 2011-06-24 13:29:48.124434 [DEBUG] switch_ivr_play_say.c:1244 Codec
> Activated L16 at 8000hz 1 channels 20ms
> 2011-06-24 13:29:48.124434 [DEBUG] switch_ivr_play_say.c:1581 done playing
> file
> 2011-06-24 13:29:48.124434 [DEBUG] switch_core_state_machine.c:366
> (sofia/external/03996563750911 at 203.208.207.212) State EXECUTE going to
> sleep
> 2011-06-24 13:29:48.124434 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996563750911 at 203.208.207.212) Running State Change
> CS_HIBERNATE
> 2011-06-24 13:29:48.124434 [DEBUG] switch_core_state_machine.c:381
> (sofia/external/03996563750911 at 203.208.207.212) State HIBERNATE
> 2011-06-24 13:29:48.124434 [DEBUG] mod_sofia.c:221 sofia/external/
> 03996563750911 at 203.208.207.212 SOFIA HIBERNATE
> 2011-06-24 13:29:48.124434 [DEBUG] switch_ivr_bridge.c:731 (sofia/external/
> 03996563750911 at 203.208.207.212) State Change CS_HIBERNATE -> CS_RESET
> 2011-06-24 13:29:48.124434 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996563750911 at 203.208.207.212 [BREAK]
> 2011-06-24 13:29:48.124434 [DEBUG] switch_core_state_machine.c:381
> (sofia/external/03996563750911 at 203.208.207.212) State HIBERNATE going to
> sleep
> 2011-06-24 13:29:48.124434 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996563750911 at 203.208.207.212) Running State Change
> CS_RESET
> 2011-06-24 13:29:48.124434 [DEBUG] switch_core_state_machine.c:362
> (sofia/external/03996563750911 at 203.208.207.212) State RESET
> 2011-06-24 13:29:48.124434 [DEBUG] mod_sofia.c:165 sofia/external/
> 03996563750911 at 203.208.207.212 SOFIA RESET
> 2011-06-24 13:29:48.124434 [DEBUG] switch_ivr_bridge.c:716 sofia/external/
> 03996563750911 at 203.208.207.212 CUSTOM RESET
> 2011-06-24 13:29:48.124434 [DEBUG] switch_core_state_machine.c:66
> sofia/external/03996563750911 at 203.208.207.212 Standard RESET
> 2011-06-24 13:29:48.124434 [DEBUG] switch_core_state_machine.c:362
> (sofia/external/03996563750911 at 203.208.207.212) State RESET going to sleep
> 2011-06-24 13:29:48.126015 [DEBUG] switch_core_session.c:709 Send signal
> sofia/external/03996563750911 at 203.208.207.212 [BREAK]
> 2011-06-24 13:29:48.142264 [DEBUG] switch_ivr_play_say.c:1581 done playing
> file
> 2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:366
> (sofia/external/03996597632298 at 203.208.207.212) State EXECUTE going to
> sleep
> 2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996597632298 at 203.208.207.212) Running State Change
> CS_HIBERNATE
> 2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:381
> (sofia/external/03996597632298 at 203.208.207.212) State HIBERNATE
> 2011-06-24 13:29:48.142264 [DEBUG] mod_sofia.c:221 sofia/external/
> 03996597632298 at 203.208.207.212 SOFIA HIBERNATE
> 2011-06-24 13:29:48.142264 [DEBUG] switch_ivr_bridge.c:731 (sofia/external/
> 03996597632298 at 203.208.207.212) State Change CS_HIBERNATE -> CS_RESET
> 2011-06-24 13:29:48.142264 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996597632298 at 203.208.207.212 [BREAK]
> 2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:381
> (sofia/external/03996597632298 at 203.208.207.212) State HIBERNATE going to
> sleep
> 2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996597632298 at 203.208.207.212) Running State Change
> CS_RESET
> 2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:362
> (sofia/external/03996597632298 at 203.208.207.212) State RESET
> 2011-06-24 13:29:48.142264 [DEBUG] mod_sofia.c:165 sofia/external/
> 03996597632298 at 203.208.207.212 SOFIA RESET
> 2011-06-24 13:29:48.142264 [DEBUG] switch_ivr_bridge.c:716 sofia/external/
> 03996597632298 at 203.208.207.212 CUSTOM RESET
> 2011-06-24 13:29:48.142264 [DEBUG] switch_ivr_bridge.c:723 (sofia/external/
> 03996597632298 at 203.208.207.212) State Change CS_RESET -> CS_SOFT_EXECUTE
> 2011-06-24 13:29:48.142264 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996597632298 at 203.208.207.212 [BREAK]
> 2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:362
> (sofia/external/03996597632298 at 203.208.207.212) State RESET going to sleep
> 2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996597632298 at 203.208.207.212) Running State Change
> CS_SOFT_EXECUTE
> 2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:372
> (sofia/external/03996597632298 at 203.208.207.212) State SOFT_EXECUTE
> 2011-06-24 13:29:48.142264 [DEBUG] mod_sofia.c:558 SOFIA SOFT_EXECUTE
> 2011-06-24 13:29:48.142264 [DEBUG] switch_ivr_bridge.c:741 sofia/external/
> 03996597632298 at 203.208.207.212 CUSTOM SOFT_EXECUTE
> 2011-06-24 13:29:48.142264 [DEBUG] switch_ivr_bridge.c:761 (sofia/external/
> 03996563750911 at 203.208.207.212) State Change CS_RESET -> CS_SOFT_EXECUTE
> 2011-06-24 13:29:48.142264 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996563750911 at 203.208.207.212 [BREAK]
> 2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996563750911 at 203.208.207.212) Running State Change
> CS_SOFT_EXECUTE
> 2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:372
> (sofia/external/03996563750911 at 203.208.207.212) State SOFT_EXECUTE
> 2011-06-24 13:29:48.142264 [DEBUG] mod_sofia.c:558 SOFIA SOFT_EXECUTE
> 2011-06-24 13:29:48.142264 [DEBUG] switch_ivr_bridge.c:741 sofia/external/
> 03996563750911 at 203.208.207.212 CUSTOM SOFT_EXECUTE
> 2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:204
> sofia/external/03996563750911 at 203.208.207.212 Standard SOFT_EXECUTE
> 2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:372
> (sofia/external/03996563750911 at 203.208.207.212) State SOFT_EXECUTE going
> to sleep
> 2011-06-24 13:29:48.161779 [DEBUG] switch_channel.c:2535 (sofia/external/
> 03996563750911 at 203.208.207.212) Callstate Change EARLY -> HANGUP
> 2011-06-24 13:29:48.161779 [NOTICE] switch_ivr_originate.c:1045 Hangup
> sofia/external/03996563750911 at 203.208.207.212 [CS_SOFT_EXECUTE]
> [DESTINATION_OUT_OF_ORDER]
> 2011-06-24 13:29:48.161779 [DEBUG] switch_channel.c:2551 Send signal
> sofia/external/03996563750911 at 203.208.207.212 [KILL]
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996563750911 at 203.208.207.212 [BREAK]
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996563750911 at 203.208.207.212) Running State Change
> CS_HANGUP
> 2011-06-24 13:29:48.161779 [DEBUG] switch_channel.c:2535 (sofia/external/
> 03996597632298 at 203.208.207.212) Callstate Change ACTIVE -> HANGUP
> 2011-06-24 13:29:48.161779 [NOTICE] switch_ivr_bridge.c:772 Hangup
> sofia/external/03996597632298 at 203.208.207.212 [CS_SOFT_EXECUTE]
> [ORIGINATOR_CANCEL]
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:557
> (sofia/external/03996563750911 at 203.208.207.212) State HANGUP
> 2011-06-24 13:29:48.161779 [DEBUG] mod_sofia.c:457 Channel sofia/external/
> 03996563750911 at 203.208.207.212 hanging up, cause: DESTINATION_OUT_OF_ORDER
> 2011-06-24 13:29:48.161779 [DEBUG] mod_sofia.c:510 Sending CANCEL to
> sofia/external/03996563750911 at 203.208.207.212
> 2011-06-24 13:29:48.161779 [DEBUG] switch_channel.c:2551 Send signal
> sofia/external/03996597632298 at 203.208.207.212 [KILL]
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996597632298 at 203.208.207.212 [BREAK]
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:372
> (sofia/external/03996597632298 at 203.208.207.212) State SOFT_EXECUTE going
> to sleep
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996597632298 at 203.208.207.212) Running State Change
> CS_HANGUP
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:46
> sofia/external/03996563750911 at 203.208.207.212 Standard HANGUP, cause:
> DESTINATION_OUT_OF_ORDER
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:557
> (sofia/external/03996563750911 at 203.208.207.212) State HANGUP going to
> sleep
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:351
> (sofia/external/03996563750911 at 203.208.207.212) State Change CS_HANGUP ->
> CS_REPORTING
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996563750911 at 203.208.207.212 [BREAK]
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996563750911 at 203.208.207.212) Running State Change
> CS_REPORTING
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:617
> (sofia/external/03996563750911 at 203.208.207.212) State REPORTING
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:53
> sofia/external/03996563750911 at 203.208.207.212 Standard REPORTING, cause:
> DESTINATION_OUT_OF_ORDER
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:617
> (sofia/external/03996563750911 at 203.208.207.212) State REPORTING going to
> sleep
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:557
> (sofia/external/03996597632298 at 203.208.207.212) State HANGUP
> 2011-06-24 13:29:48.161779 [DEBUG] mod_sofia.c:457 Channel sofia/external/
> 03996597632298 at 203.208.207.212 hanging up, cause: ORIGINATOR_CANCEL
> 2011-06-24 13:29:48.161779 [DEBUG] mod_sofia.c:500 Sending BYE to
> sofia/external/03996597632298 at 203.208.207.212
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:345
> (sofia/external/03996563750911 at 203.208.207.212) State Change CS_REPORTING
> -> CS_DESTROY
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996563750911 at 203.208.207.212 [BREAK]
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_session.c:1288 Session 52
> (sofia/external/03996563750911 at 203.208.207.212) Locked, Waiting on
> external entities
> 2011-06-24 13:29:48.161779 [NOTICE] switch_core_session.c:1306 Session 52
> (sofia/external/03996563750911 at 203.208.207.212) Ended
> 2011-06-24 13:29:48.161779 [NOTICE] switch_core_session.c:1308 Close
> Channel sofia/external/03996563750911 at 203.208.207.212 [CS_DESTROY]
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:449
> (sofia/external/03996563750911 at 203.208.207.212) Callstate Change HANGUP ->
> DOWN
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:452
> (sofia/external/03996563750911 at 203.208.207.212) Running State Change
> CS_DESTROY
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:462
> (sofia/external/03996563750911 at 203.208.207.212) State DESTROY
> 2011-06-24 13:29:48.161779 [DEBUG] mod_sofia.c:362 sofia/external/
> 03996563750911 at 203.208.207.212 SOFIA DESTROY
> 2011-06-24 13:29:48.161779 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> 0x2aaaac03fff8 (nil)
> 2011-06-24 13:29:48.161779 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
> 0x2aaaac03fff8 (nil)
> 2011-06-24 13:29:48.161779 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> 0x2aaaac068fc8 (nil)
> 2011-06-24 13:29:48.161779 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
> 0x2aaaac068fc8 (nil)
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:46
> sofia/external/03996597632298 at 203.208.207.212 Standard HANGUP, cause:
> ORIGINATOR_CANCEL
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:60
> sofia/external/03996563750911 at 203.208.207.212 Standard DESTROY
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:557
> (sofia/external/03996597632298 at 203.208.207.212) State HANGUP going to
> sleep
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:462
> (sofia/external/03996563750911 at 203.208.207.212) State DESTROY going to
> sleep
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:351
> (sofia/external/03996597632298 at 203.208.207.212) State Change CS_HANGUP ->
> CS_REPORTING
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996597632298 at 203.208.207.212 [BREAK]
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/03996597632298 at 203.208.207.212) Running State Change
> CS_REPORTING
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:617
> (sofia/external/03996597632298 at 203.208.207.212) State REPORTING
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:53
> sofia/external/03996597632298 at 203.208.207.212 Standard REPORTING, cause:
> ORIGINATOR_CANCEL
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:617
> (sofia/external/03996597632298 at 203.208.207.212) State REPORTING going to
> sleep
> send 390 bytes to udp/[203.208.207.212]:5060 at 05:29:48.671783:
>    ------------------------------------------------------------------------
>    CANCEL sip:03996563750911 at 203.208.207.212 SIP/2.0
>    Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKytFt9mX5HQNSj
>    Max-Forwards: 70
>    From: "" <sip:03996597632298 at 202.73.56.46>;tag=62eBXSt93tr7F
>    To: <sip:03996563750911 at 203.208.207.212>
>    Call-ID: d0f42376-18c5-122f-3387-0015c5fc7ad9
>    CSeq: 14110629 CANCEL
>    Reason: Q.850;cause=27;text="DESTINATION_OUT_OF_ORDER"
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:345
> (sofia/external/03996597632298 at 203.208.207.212) State Change CS_REPORTING
> -> CS_DESTROY
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/external/03996597632298 at 203.208.207.212 [BREAK]
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_session.c:1288 Session 51
> (sofia/external/03996597632298 at 203.208.207.212) Locked, Waiting on
> external entities
> 2011-06-24 13:29:48.161779 [NOTICE] switch_core_session.c:1306 Session 51
> (sofia/external/03996597632298 at 203.208.207.212) Ended
> 2011-06-24 13:29:48.161779 [NOTICE] switch_core_session.c:1308 Close
> Channel sofia/external/03996597632298 at 203.208.207.212 [CS_DESTROY]
> send 677 bytes to udp/[203.208.207.212]:5060 at 05:29:48.671936:
>    ------------------------------------------------------------------------
>    BYE sip:203.208.207.212:5060 SIP/2.0
>    Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bK0c2BDBZcc91yS
>    Max-Forwards: 70
>    From: "" <sip:03996563750911 at 202.73.56.46>;tag=5SNjUy956H2mm
>    To: <sip:03996597632298 at 203.208.207.212
> >;tag=2QGB951HCR30000E1D0001Ll000B8RX0S6R1TX
>    Call-ID: ca5d29b7-18c5-122f-3387-0015c5fc7ad9
>    CSeq: 14110627 BYE
>    Contact: <sip:mod_sofia at 202.73.56.46:5080>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, PRACK, NOTIFY
>    Supported: 100rel, timer, precondition, path, replaces
>    Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL"
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:449
> (sofia/external/03996597632298 at 203.208.207.212) Callstate Change HANGUP ->
> DOWN
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:452
> (sofia/external/03996597632298 at 203.208.207.212) Running State Change
> CS_DESTROY
> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:462
> (sofia/external/03996597632298 at 203.208.207.212) State DESTROY
> 2011-06-24 13:29:48.161779 [DEBUG] mod_sofia.c:362 sofia/external/
> 03996597632298 at 203.208.207.212 SOFIA DESTROY
> 2011-06-24 13:29:48.161779 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> 0x2aaab8030878 (nil)
> 2011-06-24 13:29:48.161779 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
> 0x2aaab8030878 (nil)
> 2011-06-24 13:29:48.161779 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> 0x2aaab80894b8 0x5112110
> 2011-06-24 13:29:48.161779 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
> 0x2aaab80894b8 (nil)
> 2011-06-24 13:29:48.166430 [INFO] mod_com_g729.c:83 ENCODER DESTROY -
> 0x2aaab80894b8 0x5112110
> 2011-06-24 13:29:48.166430 [DEBUG] switch_core_state_machine.c:60
> sofia/external/03996597632298 at 203.208.207.212 Standard DESTROY
> 2011-06-24 13:29:48.166430 [DEBUG] switch_core_state_machine.c:462
> (sofia/external/03996597632298 at 203.208.207.212) State DESTROY going to
> sleep
> recv 383 bytes from udp/[203.208.207.212]:5060 at 05:29:48.679561:
>    ------------------------------------------------------------------------
>    SIP/2.0 200 Ok
>    Via: SIP/2.0/UDP 10.1.1.46:5080
> ;rport=5080;branch=z9hG4bKytFt9mX5HQNSj;received=10.1.1.46
>    From: "" <sip:03996597632298 at 10.1.1.46>;tag=62eBXSt93tr7F
>    To: <sip:03996563750911 at 203.208.207.212
> >;tag=2QGB951HCR30000E1D00000u000000013MCNME
>    Call-ID: d0f42376-18c5-122f-3387-0015c5fc7ad9
>    CSeq: 14110629 CANCEL
>    Contact: <sip:203.208.207.212:5060>
>    Content-Length:     0
>
>    ------------------------------------------------------------------------
> recv 731 bytes from udp/[203.208.207.212]:5060 at 05:29:48.681395:
>    ------------------------------------------------------------------------
>    UPDATE sip:mod_sofia at 10.1.1.46:5080 SIP/2.0
>    Via: SIP/2.0/UDP 203.208.207.212:5060
> ;branch=z9hG4bK00151746C47A8307FCE25ED9D752
>    From: <sip:03996563750911 at 203.208.207.212
> >;tag=2QGB951HCR30000E1D0001Ll000B8RV0KORLQK
>    To: "" <sip:03996597632298 at 10.1.1.46>;tag=62eBXSt93tr7F
>    Call-ID: d0f42376-18c5-122f-3387-0015c5fc7ad9
>    CSeq: 46496 UPDATE
>    Contact: <sip:03996563750911 at 203.208.207.212:5060>
>    Content-Type: application/sdp
>    Max-Forwards: 70
>    Supported: 100rel, timer, replaces
>    Content-Length:   231
>
>    v=0
>    o=- 508550174225779858 2 IN IP4 203.208.207.212
>    s=session
>    c=IN IP4 203.208.207.202
>    t=0 0
>    m=audio 25150 RTP/AVP 18 101
>    a=rtpmap:18 G729/8000
>    a=fmtp:18 annexb=no
>    a=rtpmap:101 telephone-event/8000
>    a=ptime:20
>    a=inactive
>    ------------------------------------------------------------------------
> send 895 bytes to udp/[203.208.207.212]:5060 at 05:29:48.681661:
>    ------------------------------------------------------------------------
>    SIP/2.0 200 OK
>    Via: SIP/2.0/UDP 203.208.207.212:5060
> ;branch=z9hG4bK00151746C47A8307FCE25ED9D752
>    From: <sip:03996563750911 at 203.208.207.212
> >;tag=2QGB951HCR30000E1D0001Ll000B8RV0KORLQK
>    To: "" <sip:03996597632298 at 10.1.1.46>;tag=62eBXSt93tr7F
>    Call-ID: d0f42376-18c5-122f-3387-0015c5fc7ad9
>    CSeq: 46496 UPDATE
>    Contact: <sip:mod_sofia at 202.73.56.46:5080>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, PRACK, NOTIFY
>    Supported: 100rel, timer, precondition, path, replaces
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 216
>
>    v=0
>    o=FreeSWITCH 1308872756 1308872758 IN IP4 202.73.56.46
>    s=FreeSWITCH
>    c=IN IP4 202.73.56.46
>    t=0 0
>    m=audio 20630 RTP/AVP 18 101 3 13
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=inactive
>    a=ptime:20
>    ------------------------------------------------------------------------
> recv 411 bytes from udp/[203.208.207.212]:5060 at 05:29:48.681824:
>    ------------------------------------------------------------------------
>    SIP/2.0 487 Request Terminated
>    Via: SIP/2.0/UDP 10.1.1.46:5080
> ;rport=5080;branch=z9hG4bKytFt9mX5HQNSj;received=10.1.1.46
>    From: "" <sip:03996597632298 at 10.1.1.46>;tag=62eBXSt93tr7F
>    To: <sip:03996563750911 at 203.208.207.212
> >;tag=2QGB951HCR30000E1D0001Ll000B8RV0KORLQK
>    Call-ID: d0f42376-18c5-122f-3387-0015c5fc7ad9
>    CSeq: 14110629 INVITE
>    Reason: SIP;cause=487;text="Request Terminated"
>    Content-Length:     0
>
>    ------------------------------------------------------------------------
> send 371 bytes to udp/[203.208.207.212]:5060 at 05:29:48.681913:
>    ------------------------------------------------------------------------
>    ACK sip:03996563750911 at 203.208.207.212 SIP/2.0
>    Via: SIP/2.0/UDP 202.73.56.46:5080;rport;branch=z9hG4bKytFt9mX5HQNSj
>    Max-Forwards: 70
>    From: "" <sip:03996597632298 at 202.73.56.46>;tag=62eBXSt93tr7F
>    To: <sip:03996563750911 at 203.208.207.212
> >;tag=2QGB951HCR30000E1D0001Ll000B8RV0KORLQK
>    Call-ID: d0f42376-18c5-122f-3387-0015c5fc7ad9
>    CSeq: 14110629 ACK
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> recv 380 bytes from udp/[203.208.207.212]:5060 at 05:29:48.683041:
>    ------------------------------------------------------------------------
>    SIP/2.0 200 Ok
>    Via: SIP/2.0/UDP 10.1.1.46:5080
> ;rport=5080;branch=z9hG4bK0c2BDBZcc91yS;received=10.1.1.46
>    From: "" <sip:03996563750911 at 10.1.1.46>;tag=5SNjUy956H2mm
>    To: <sip:03996597632298 at 203.208.207.212
> >;tag=2QGB951HCR30000E1D0001Ll000B8RX0S6R1TX
>    Call-ID: ca5d29b7-18c5-122f-3387-0015c5fc7ad9
>    CSeq: 14110627 BYE
>    Contact: <sip:203.208.207.212:5060>
>    Content-Length:     0
>
>    ------------------------------------------------------------------------
> 2011-06-24 13:29:48.524051 [DEBUG] switch_scheduler.c:138 Deleting task 27
> switch_ivr_schedule_hangup (c0bd700d-913c-42ad-b68f-81001bf658b8)
>
>
>
>
>
>
> On Fri, Jun 24, 2011 at 2:41 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Pastebin the entire debug log, including the siptrace. Also include the
>> originate line and any other dialplan config that might be used.
>> -MC
>>
>>
>> On Thu, Jun 23, 2011 at 11:26 PM, David Ma <mays.david at gmail.com> wrote:
>>
>>> Hi Michael,
>>>
>>> Unfortunately this problem still happens.
>>>
>>> I enabled "continue_on_fail" for leg-A when I originated the call. Leg-A
>>> call went well. Then I originated leg-B call ("continue_on_fail" is NOT set
>>> for leg-B), which failed for [DESTINATION_OUT_OF_ORDER]. As the consequence,
>>> leg-A was hung up by FS automatically for [ORIGINATOR_CANCEL].
>>>
>>> The log excerpt follows.
>>>
>>> Do you think "continue_on_fail" should be also enabled for leg-B call?
>>>
>>> Thanks,
>>> D.Ma
>>>
>>> 2011-06-24 13:29:35.887830 [DEBUG] switch_ivr_originate.c:1971 variable
>>> string 5 = [continue_on_fail=true]
>>>
>>> 2011-06-24 13:29:35.887830 [NOTICE] switch_channel.c:808 New Channel
>>> sofia/external/03996597632298 at 203.208.207.212[c0bd700d-913c-42ad-b68f-81001bf658b8]
>>> [...]
>>> 2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:372
>>> (sofia/external/03996563750911 at 203.208.207.212) State SOFT_EXECUTE going
>>> to sleep
>>> 2011-06-24 13:29:48.161779 [DEBUG] switch_channel.c:2535 (sofia/external/
>>> 03996563750911 at 203.208.207.212) Callstate Change EARLY -> HANGUP
>>> 2011-06-24 13:29:48.161779 [NOTICE] switch_ivr_originate.c:1045 Hangup
>>> sofia/external/03996563750911 at 203.208.207.212 [CS_SOFT_EXECUTE]
>>> [DESTINATION_OUT_OF_ORDER]
>>> 2011-06-24 13:29:48.161779 [DEBUG] switch_channel.c:2551 Send signal
>>> sofia/external/03996563750911 at 203.208.207.212 [KILL]
>>> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_session.c:1116 Send signal
>>> sofia/external/03996563750911 at 203.208.207.212 [BREAK]
>>> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:320
>>> (sofia/external/03996563750911 at 203.208.207.212) Running State Change
>>> CS_HANGUP
>>> 2011-06-24 13:29:48.161779 [DEBUG] switch_channel.c:2535 (sofia/external/
>>> 03996597632298 at 203.208.207.212) Callstate Change ACTIVE -> HANGUP
>>> 2011-06-24 13:29:48.161779 [NOTICE] switch_ivr_bridge.c:772 Hangup
>>> sofia/external/03996597632298 at 203.208.207.212 [CS_SOFT_EXECUTE]
>>> [ORIGINATOR_CANCEL]
>>> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:557
>>> (sofia/external/03996563750911 at 203.208.207.212) State HANGUP
>>> 2011-06-24 13:29:48.161779 [DEBUG] mod_sofia.c:457 Channel
>>> sofia/external/03996563750911 at 203.208.207.212 hanging up, cause:
>>> DESTINATION_OUT_OF_ORDER
>>> 2011-06-24 13:29:48.161779 [DEBUG] mod_sofia.c:510 Sending CANCEL to
>>> sofia/external/03996563750911 at 203.208.207.212
>>> 2011-06-24 13:29:48.161779 [DEBUG] switch_channel.c:2551 Send signal
>>> sofia/external/03996597632298 at 203.208.207.212 [KILL]
>>> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_session.c:1116 Send signal
>>> sofia/external/03996597632298 at 203.208.207.212 [BREAK]
>>>
>>> On Fri, Jun 17, 2011 at 10:52 AM, David Ma <mays.david at gmail.com> wrote:
>>>
>>>> Hi Michael,
>>>>
>>>> Thanks very much for your prompt response! I appreciate the information
>>>> provided.
>>>>
>>>> I was actually searching the the existence of such a variable. I was not
>>>> so luck to find it out and thereby resort to the support forum.
>>>>
>>>> I've modified my code to build this parameter into my application. Will
>>>> feedback to you after verification.
>>>>
>>>> Thanks again,
>>>> D.Ma
>>>>
>>>> On Fri, Jun 17, 2011 at 4:51 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>> How about setting this?
>>>>> http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
>>>>>
>>>>> -MC
>>>>>
>>>>>
>>>>> On Thu, Jun 16, 2011 at 1:32 AM, dma <mays.david at gmail.com> wrote:
>>>>>
>>>>>> I am creating a call-back solution. After leg-A answers, I originate
>>>>>> leg-B
>>>>>> call. After receiving SIP 183 from Leg-B, I bridge the 2 legs.
>>>>>> However, in
>>>>>> some cases, leg-A is automatically disconnected by FreeSwitch on leg-B
>>>>>> failure, for example, DESTINATION_OUT_OF_ORDER. The application is not
>>>>>> given
>>>>>> a chance to handle leg-B failure event. This should not be a correct
>>>>>> scenario because I never set "hangup-after-bridge", which is false by
>>>>>> default.
>>>>>>
>>>>>> The right way should be, FreeSwitch doesn't hang up leg-A
>>>>>> automatically, but
>>>>>> give a chance for the application to decide what to do.
>>>>>>
>>>>>> Please see the logs below:
>>>>>>
>>>>>> =================================================
>>>>>>
>>>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971
>>>>>> variable
>>>>>> string 0 = [origination_caller_id_number=03996563750914]
>>>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971
>>>>>> variable
>>>>>> string 1 = [originate_timeout=30]
>>>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971
>>>>>> variable
>>>>>> string 2 = [ccd_session_id=20110610105829676824]
>>>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971
>>>>>> variable
>>>>>> string 3 = [sip_cid_type=pid]
>>>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971
>>>>>> variable
>>>>>> string 4 = [privacy=yes]
>>>>>> 2011-06-10 11:09:55.370506 [NOTICE] switch_channel.c:808 New Channel
>>>>>> sofia/external/03996590031055 at 203.208.207.212
>>>>>> [ea57b74b-a8c2-4fea-9683-98054dc03a79]
>>>>>> 2011-06-10 11:09:55.370506 [DEBUG] mod_sofia.c:4129
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_NEW
>>>>>> ->
>>>>>> CS_INIT
>>>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_core_session.c:1116 Send
>>>>>> signal
>>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_core_state_machine.c:320
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>>> CS_INIT
>>>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_core_state_machine.c:356
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State INIT
>>>>>> 2011-06-10 11:09:55.370506 [DEBUG] mod_sofia.c:84
>>>>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA INIT
>>>>>> send 984 bytes to udp/[203.208.207.212]:5060 at 03:09:55.477997:
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>>   INVITE sip:03996590031055 at 203.208.207.212 SIP/2.0
>>>>>>   Via: SIP/2.0/UDP 202.73.56.46:5080
>>>>>> ;rport;branch=z9hG4bKXjQ7eFpKypy5D
>>>>>>   Max-Forwards: 70
>>>>>>   From: "" &lt;sip:03996563750914 at 202.73.56.46&gt;;tag=H4ZN43UH9S9Qc
>>>>>>   To: &lt;sip:03996590031055 at 203.208.207.212&gt;
>>>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>>>   CSeq: 13501633 INVITE
>>>>>>   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
>>>>>>   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
>>>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>>>> REGISTER, REFER, NOTIFY
>>>>>>   Supported: timer, precondition, path, replaces
>>>>>>   Allow-Events: talk, hold, refer
>>>>>>   Privacy: none
>>>>>>   Content-Type: application/sdp
>>>>>>   Content-Disposition: session
>>>>>>   Content-Length: 204
>>>>>>   X-FS-Support: update_display
>>>>>>   P-Asserted-Identity: "" &lt;sip:03996563750914 at 202.73.56.46&gt;
>>>>>>
>>>>>>   v=0
>>>>>>   o=FreeSWITCH 1307645395 1307645396 IN IP4 202.73.56.46
>>>>>>   s=FreeSWITCH
>>>>>>   c=IN IP4 202.73.56.46
>>>>>>   t=0 0
>>>>>>   m=audio 30000 RTP/AVP 18 3 101 13
>>>>>>   a=rtpmap:101 telephone-event/8000
>>>>>>   a=fmtp:101 0-16
>>>>>>   a=ptime:20
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>> 2011-06-10 11:09:55.371674 [DEBUG] mod_sofia.c:124
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_INIT
>>>>>> ->
>>>>>> CS_ROUTING
>>>>>> 2011-06-10 11:09:55.371674 [DEBUG] switch_core_session.c:1116 Send
>>>>>> signal
>>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>>> 2011-06-10 11:09:55.371674 [DEBUG] sofia.c:4646 Channel
>>>>>> sofia/external/03996590031055 at 203.208.207.212 entering state
>>>>>> [calling][0]
>>>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:356
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State INIT going to
>>>>>> sleep
>>>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:320
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>>> CS_ROUTING
>>>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_channel.c:1657
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change DOWN
>>>>>> ->
>>>>>> RINGING
>>>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:359
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State ROUTING
>>>>>> 2011-06-10 11:09:55.373478 [DEBUG] mod_sofia.c:147
>>>>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA ROUTING
>>>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_ivr_originate.c:66
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change
>>>>>> CS_ROUTING ->
>>>>>> CS_CONSUME_MEDIA
>>>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_session.c:1116 Send
>>>>>> signal
>>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:359
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State ROUTING going
>>>>>> to sleep
>>>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:320
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>>> CS_CONSUME_MEDIA
>>>>>> 2011-06-10 11:09:55.374485 [DEBUG] switch_core_state_machine.c:378
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State CONSUME_MEDIA
>>>>>> 2011-06-10 11:09:55.374485 [DEBUG] switch_core_state_machine.c:378
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State CONSUME_MEDIA
>>>>>> going to
>>>>>> sleep
>>>>>> recv 307 bytes from udp/[203.208.207.212]:5060 at 03:09:55.485130:
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>>   SIP/2.0 100 Trying
>>>>>>   Via: SIP/2.0/UDP
>>>>>> 10.1.1.46:5080
>>>>>> ;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>>>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>>>>   To: &lt;sip:03996590031055 at 203.208.207.212&gt;
>>>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>>>   CSeq: 13501633 INVITE
>>>>>>   Content-Length:     0
>>>>>>
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>> recv 669 bytes from udp/[203.208.207.212]:5060 at 03:09:56.677788:
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>>   SIP/2.0 183 Session Progress
>>>>>>   Via: SIP/2.0/UDP
>>>>>> 10.1.1.46:5080
>>>>>> ;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>>>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>>>>   To:
>>>>>> &lt;sip:03996590031055 at 203.208.207.212
>>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>>>   CSeq: 13501633 INVITE
>>>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER,
>>>>>> NOTIFY,
>>>>>> SUBSCRIBE, UPDATE
>>>>>>   Content-Type: application/sdp
>>>>>>   Content-Length:   189
>>>>>>
>>>>>>   v=0
>>>>>>   o=- 421265648 1 IN IP4 203.208.207.219
>>>>>>   s=session
>>>>>>   c=IN IP4 203.208.207.196
>>>>>>   t=0 0
>>>>>>   m=audio 30792 RTP/AVP 18 101
>>>>>>   a=rtpmap:18 G729/8000
>>>>>>   a=rtpmap:101 telephone-event/8000
>>>>>>   a=sendrecv
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>> 2011-06-10 11:09:56.571839 [INFO] sofia.c:729
>>>>>> sofia/external/03996590031055 at 203.208.207.212 Update Callee ID to
>>>>>> "Outbound
>>>>>> Call" <03996590031055>
>>>>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia.c:4646 Channel
>>>>>> sofia/external/03996590031055 at 203.208.207.212 entering state
>>>>>> [proceeding][183]
>>>>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia.c:4657 Remote SDP:
>>>>>> v=0
>>>>>> o=- 421265648 1 IN IP4 203.208.207.219
>>>>>> s=session
>>>>>> c=IN IP4 203.208.207.196
>>>>>> t=0 0
>>>>>> m=audio 30792 RTP/AVP 18 101
>>>>>> a=rtpmap:18 G729/8000
>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>
>>>>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:4467 Audio Codec
>>>>>> Compare
>>>>>> [G729:18:8000:20:8000]/[G729:18:8000:20:8000]
>>>>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:2757 Set Codec
>>>>>> sofia/external/03996590031055 at 203.208.207.212 G729/8000 20 ms 160
>>>>>> samples
>>>>>> 8000 bits
>>>>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:4565 Set 2833 dtmf
>>>>>> send
>>>>>> payload to 101
>>>>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:2987 AUDIO RTP
>>>>>> [sofia/external/03996590031055 at 203.208.207.212] 10.1.1.46 port 30000
>>>>>> ->
>>>>>> 203.208.207.196 port 30792 codec: 18 ms: 20
>>>>>> 2011-06-10 11:09:56.571839 [DEBUG] switch_rtp.c:1607 Starting timer
>>>>>> [soft]
>>>>>> 160 bytes per 20ms
>>>>>> 2011-06-10 11:09:56.573731 [DEBUG] sofia_glue.c:3228 Set 2833 dtmf
>>>>>> send
>>>>>> payload to 101
>>>>>> 2011-06-10 11:09:56.573731 [DEBUG] sofia_glue.c:3233 Set 2833 dtmf
>>>>>> receive
>>>>>> payload to 101
>>>>>> 2011-06-10 11:09:56.573731 [NOTICE] sofia_glue.c:3680 Pre-Answer
>>>>>> sofia/external/03996590031055 at 203.208.207.212!
>>>>>> 2011-06-10 11:09:56.573731 [DEBUG] switch_channel.c:2627
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change
>>>>>> RINGING ->
>>>>>> EARLY
>>>>>> 2011-06-10 11:09:56.573731 [DEBUG] switch_ivr_originate.c:3408
>>>>>> Originate
>>>>>> Resulted in Success: [sofia/external/03996590031055 at 203.208.207.212]
>>>>>> 2011-06-10 11:09:56.573731 [DEBUG] mod_commands.c:3205
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change
>>>>>> CS_CONSUME_MEDIA -> CS_EXECUTE
>>>>>> 2011-06-10 11:09:56.573731 [DEBUG] switch_core_session.c:1116 Send
>>>>>> signal
>>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>>> 2011-06-10 11:09:56.575262 [DEBUG] switch_core_state_machine.c:320
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>>> CS_EXECUTE
>>>>>> 2011-06-10 11:09:56.575262 [DEBUG] switch_core_state_machine.c:366
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State EXECUTE
>>>>>> 2011-06-10 11:09:56.575262 [DEBUG] mod_sofia.c:240
>>>>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA EXECUTE
>>>>>> 2011-06-10 11:09:56.575262 [DEBUG] switch_core_state_machine.c:157
>>>>>> sofia/external/03996590031055 at 203.208.207.212 Standard EXECUTE
>>>>>> EXECUTE sofia/external/03996590031055 at 203.208.207.212 park()
>>>>>> 2011-06-10 11:09:56.618770 [DEBUG] switch_rtp.c:2933 Correct ip/port
>>>>>> confirmed.
>>>>>> recv 669 bytes from udp/[203.208.207.212]:5060 at 03:10:01.562021:
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>>   SIP/2.0 183 Session Progress
>>>>>>   Via: SIP/2.0/UDP
>>>>>> 10.1.1.46:5080
>>>>>> ;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>>>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>>>>   To:
>>>>>> &lt;sip:03996590031055 at 203.208.207.212
>>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>>>   CSeq: 13501633 INVITE
>>>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER,
>>>>>> NOTIFY,
>>>>>> SUBSCRIBE, UPDATE
>>>>>>   Content-Type: application/sdp
>>>>>>   Content-Length:   189
>>>>>>
>>>>>>   v=0
>>>>>>   o=- 421265648 2 IN IP4 203.208.207.219
>>>>>>   s=session
>>>>>>   c=IN IP4 203.208.207.196
>>>>>>   t=0 0
>>>>>>   m=audio 30792 RTP/AVP 18 101
>>>>>>   a=rtpmap:18 G729/8000
>>>>>>   a=rtpmap:101 telephone-event/8000
>>>>>>   a=sendrecv
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>> 2011-06-10 11:10:01.455944 [INFO] sofia.c:729
>>>>>> sofia/external/03996590031055 at 203.208.207.212 Update Callee ID to
>>>>>> "03996590031055" <03996590031055>
>>>>>> 2011-06-10 11:10:01.455944 [DEBUG] sofia.c:4641 Channel
>>>>>> sofia/external/03996590031055 at 203.208.207.212 skipping state
>>>>>> [proceeding][183]
>>>>>> recv 669 bytes from udp/[203.208.207.212]:5060 at 03:10:01.563178:
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>>   SIP/2.0 183 Session Progress
>>>>>>   Via: SIP/2.0/UDP
>>>>>> 10.1.1.46:5080
>>>>>> ;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>>>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>>>>   To:
>>>>>> &lt;sip:03996590031055 at 203.208.207.212
>>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>>>   CSeq: 13501633 INVITE
>>>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER,
>>>>>> NOTIFY,
>>>>>> SUBSCRIBE, UPDATE
>>>>>>   Content-Type: application/sdp
>>>>>>   Content-Length:   189
>>>>>>
>>>>>>   v=0
>>>>>>   o=- 421265648 3 IN IP4 203.208.207.219
>>>>>>   s=session
>>>>>>   c=IN IP4 203.208.207.196
>>>>>>   t=0 0
>>>>>>   m=audio 30792 RTP/AVP 18 101
>>>>>>   a=rtpmap:18 G729/8000
>>>>>>   a=rtpmap:101 telephone-event/8000
>>>>>>   a=sendrecv
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>> 2011-06-10 11:10:01.457169 [DEBUG] sofia.c:4641 Channel
>>>>>> sofia/external/03996590031055 at 203.208.207.212 skipping state
>>>>>> [proceeding][183]
>>>>>> recv 749 bytes from udp/[203.208.207.212]:5060 at 03:10:27.365284:
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>>   SIP/2.0 200 Ok
>>>>>>   Via: SIP/2.0/UDP
>>>>>> 10.1.1.46:5080
>>>>>> ;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>>>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>>>>   To:
>>>>>> &lt;sip:03996590031055 at 203.208.207.212
>>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>>>   CSeq: 13501633 INVITE
>>>>>>   Contact: <sip:203.208.207.212:5060>
>>>>>>   Allow-Events: refer
>>>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER,
>>>>>> NOTIFY,
>>>>>> SUBSCRIBE, UPDATE
>>>>>>   Content-Type: application/sdp
>>>>>>   Supported: 100rel, timer, replaces
>>>>>>   Content-Length:   189
>>>>>>
>>>>>>   v=0
>>>>>>   o=- 421265648 4 IN IP4 203.208.207.219
>>>>>>   s=session
>>>>>>   c=IN IP4 203.208.207.196
>>>>>>   t=0 0
>>>>>>   m=audio 30792 RTP/AVP 18 101
>>>>>>   a=rtpmap:18 G729/8000
>>>>>>   a=rtpmap:101 telephone-event/8000
>>>>>>   a=sendrecv
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>> 2011-06-10 11:10:27.258092 [DEBUG] sofia.c:4646 Channel
>>>>>> sofia/external/03996590031055 at 203.208.207.212 entering state
>>>>>> [completing][200]
>>>>>> 2011-06-10 11:10:27.258092 [DEBUG] sofia.c:4657 Remote SDP:
>>>>>> v=0
>>>>>> o=- 421265648 4 IN IP4 203.208.207.219
>>>>>> s=session
>>>>>> c=IN IP4 203.208.207.196
>>>>>> t=0 0
>>>>>> m=audio 30792 RTP/AVP 18 101
>>>>>> a=rtpmap:18 G729/8000
>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>
>>>>>> send 405 bytes to udp/[203.208.207.212]:5060 at 03:10:27.366562:
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>>   ACK sip:203.208.207.212:5060 SIP/2.0
>>>>>>   Via: SIP/2.0/UDP 202.73.56.46:5080
>>>>>> ;rport;branch=z9hG4bKyUg0ga7pUZmrS
>>>>>>   Max-Forwards: 70
>>>>>>   From: "" &lt;sip:03996563750914 at 202.73.56.46&gt;;tag=H4ZN43UH9S9Qc
>>>>>>   To:
>>>>>> &lt;sip:03996590031055 at 203.208.207.212
>>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>>>   CSeq: 13501633 ACK
>>>>>>   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
>>>>>>   Content-Length: 0
>>>>>>
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>> 2011-06-10 11:10:27.260267 [DEBUG] sofia.c:4646 Channel
>>>>>> sofia/external/03996590031055 at 203.208.207.212 entering state
>>>>>> [ready][200]
>>>>>> 2011-06-10 11:10:27.260267 [DEBUG] switch_channel.c:2782
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change
>>>>>> EARLY ->
>>>>>> ACTIVE
>>>>>> 2011-06-10 11:10:27.260267 [NOTICE] sofia.c:5175 Channel
>>>>>> [sofia/external/03996590031055 at 203.208.207.212] has been answered
>>>>>> 2011-06-10 11:10:27.264178 [DEBUG] switch_scheduler.c:214 Added task
>>>>>> 23
>>>>>> switch_ivr_schedule_hangup (ea57b74b-a8c2-4fea-9683-98054dc03a79) to
>>>>>> run at
>>>>>> 1307676927
>>>>>> 2011-06-10 11:10:27.265202 [DEBUG] switch_core_session.c:954 Send
>>>>>> signal
>>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971
>>>>>> variable
>>>>>> string 0 = [origination_caller_id_number=03996590031055]
>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971
>>>>>> variable
>>>>>> string 1 = [ccd_session_id=20110610105829676824]
>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971
>>>>>> variable
>>>>>> string 2 = [sip_cid_type=pid]
>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971
>>>>>> variable
>>>>>> string 3 = [privacy=yes]
>>>>>> 2011-06-10 11:10:27.266218 [NOTICE] switch_channel.c:808 New Channel
>>>>>> sofia/external/03996563750914 at 203.208.207.212
>>>>>> [30228d2b-756a-4a98-871d-db63a2955b52]
>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] mod_sofia.c:4129
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_NEW
>>>>>> ->
>>>>>> CS_INIT
>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_session.c:1116 Send
>>>>>> signal
>>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:320
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>>> CS_INIT
>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:356
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State INIT
>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] mod_sofia.c:84
>>>>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA INIT
>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] mod_sofia.c:124
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_INIT
>>>>>> ->
>>>>>> CS_ROUTING
>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_session.c:1116 Send
>>>>>> signal
>>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:356
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State INIT going to
>>>>>> sleep
>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:320
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>>> CS_ROUTING
>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_channel.c:1657
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Callstate Change DOWN
>>>>>> ->
>>>>>> RINGING
>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:359
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State ROUTING
>>>>>> send 984 bytes to udp/[203.208.207.212]:5060 at 03:10:27.373220:
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>>   INVITE sip:03996563750914 at 203.208.207.212 SIP/2.0
>>>>>>   Via: SIP/2.0/UDP
>>>>>> 202.73.56.46:5080;rport;branch=z9hG4bKZ49rj5Qtr8aBN2011-06-10
>>>>>> 11:10:27.266218 [DEBUG] mod_sofia.c:147
>>>>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA ROUTING
>>>>>>
>>>>>>   Max-Forwards: 70
>>>>>>   From: "" &lt;sip:03996590031055 at 202.73.56.46&gt;;tag=jDSe6ycN62Zar
>>>>>>   To: &lt;sip:03996563750914 at 203.208.207.212&gt;
>>>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>>>   CSeq: 13501649 INVITE
>>>>>>   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
>>>>>>   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
>>>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>>>> REGISTER, REFER, NOTIFY
>>>>>>   Supported: timer, precondition, path, replaces
>>>>>>   Allow-Events: talk, hold, refer
>>>>>>   Privacy: none
>>>>>>   Content-Type: application/sdp
>>>>>>   Content-Disposition: session
>>>>>>   Content-Length: 204
>>>>>>   X-FS-Support: update_display
>>>>>>   P-Asserted-Identity: "" &lt;sip:03996590031055 at 202.73.56.46&gt;
>>>>>>
>>>>>>   v=0
>>>>>>   o=FreeSWITCH 1307646863 1307646864 IN IP4 202.73.56.46
>>>>>>   s=FreeSWITCH
>>>>>>   c=IN IP4 202.73.56.46
>>>>>>   t=0 0
>>>>>>   m=audio 28564 RTP/AVP 18 3 101 13
>>>>>>   a=rtpmap:101 telephone-event/8000
>>>>>>   a=fmtp:101 0-16
>>>>>>   a=ptime:20
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>> 2011-06-10 11:10:27.267630 [DEBUG] sofia.c:4646 Channel
>>>>>> sofia/external/03996563750914 at 203.208.207.212 entering state
>>>>>> [calling][0]
>>>>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_ivr_originate.c:66
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change
>>>>>> CS_ROUTING ->
>>>>>> CS_CONSUME_MEDIA
>>>>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_session.c:1116 Send
>>>>>> signal
>>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:359
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State ROUTING going
>>>>>> to sleep
>>>>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:320
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>>> CS_CONSUME_MEDIA
>>>>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:378
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State CONSUME_MEDIA
>>>>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:378
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State CONSUME_MEDIA
>>>>>> going to
>>>>>> sleep
>>>>>> recv 307 bytes from udp/[203.208.207.212]:5060 at 03:10:27.378710:
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>>   SIP/2.0 100 Trying
>>>>>>   Via: SIP/2.0/UDP
>>>>>> 10.1.1.46:5080
>>>>>> ;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>>>>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>>>>>   To: &lt;sip:03996563750914 at 203.208.207.212&gt;
>>>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>>>   CSeq: 13501649 INVITE
>>>>>>   Content-Length:     0
>>>>>>
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>> 2011-06-10 11:10:27.278484 [DEBUG] switch_ivr.c:563
>>>>>> sofia/external/03996590031055 at 203.208.207.212 Command Execute
>>>>>> playback(/usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav)
>>>>>> EXECUTE sofia/external/03996590031055 at 203.208.207.212
>>>>>> playback(/usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav)
>>>>>> 2011-06-10 11:10:27.278484 [DEBUG] switch_core_file.c:176 File
>>>>>> /usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav sample rate
>>>>>> 11025
>>>>>> doesn't match requested rate 8000
>>>>>> 2011-06-10 11:10:27.278484 [WARNING] switch_core_file.c:189 File has 2
>>>>>> channels, muxing to mono will occur.
>>>>>> 2011-06-10 11:10:27.278484 [DEBUG] switch_ivr_play_say.c:1244 Codec
>>>>>> Activated L16 at 8000hz 2 channels 20ms
>>>>>> 2011-06-10 11:10:27.298764 [INFO] mod_com_g729.c:119 ENCODER CREATE -
>>>>>> 0x2aaab00310c0 0x2aaab00b20c0
>>>>>> recv 666 bytes from udp/[203.208.207.212]:5060 at 03:10:28.618472:
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>>   SIP/2.0 183 Session Progress
>>>>>>   Via: SIP/2.0/UDP
>>>>>> 10.1.1.46:5080
>>>>>> ;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>>>>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>>>>>   To:
>>>>>> &lt;sip:03996563750914 at 203.208.207.212
>>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
>>>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>>>   CSeq: 13501649 INVITE
>>>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER,
>>>>>> NOTIFY,
>>>>>> SUBSCRIBE, UPDATE
>>>>>>   Content-Type: application/sdp
>>>>>>   Content-Length:   186
>>>>>>
>>>>>>   v=0
>>>>>>   o=- 131082 1 IN IP4 203.208.207.218
>>>>>>   s=session
>>>>>>   c=IN IP4 203.208.207.195
>>>>>>   t=0 0
>>>>>>   m=audio 45002 RTP/AVP 18 101
>>>>>>   a=rtpmap:18 G729/8000
>>>>>>   a=rtpmap:101 telephone-event/8000
>>>>>>   a=sendrecv
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>> 2011-06-10 11:10:28.513195 [INFO] sofia.c:729
>>>>>> sofia/external/03996563750914 at 203.208.207.212 Update Callee ID to
>>>>>> "Outbound
>>>>>> Call" <03996563750914>
>>>>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia.c:4646 Channel
>>>>>> sofia/external/03996563750914 at 203.208.207.212 entering state
>>>>>> [proceeding][183]
>>>>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia.c:4657 Remote SDP:
>>>>>> v=0
>>>>>> o=- 131082 1 IN IP4 203.208.207.218
>>>>>> s=session
>>>>>> c=IN IP4 203.208.207.195
>>>>>> t=0 0
>>>>>> m=audio 45002 RTP/AVP 18 101
>>>>>> a=rtpmap:18 G729/8000
>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>
>>>>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:4467 Audio Codec
>>>>>> Compare
>>>>>> [G729:18:8000:20:8000]/[G729:18:8000:20:8000]
>>>>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:2757 Set Codec
>>>>>> sofia/external/03996563750914 at 203.208.207.212 G729/8000 20 ms 160
>>>>>> samples
>>>>>> 8000 bits
>>>>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:4565 Set 2833 dtmf
>>>>>> send
>>>>>> payload to 101
>>>>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:2987 AUDIO RTP
>>>>>> [sofia/external/03996563750914 at 203.208.207.212] 10.1.1.46 port 28564
>>>>>> ->
>>>>>> 203.208.207.195 port 45002 codec: 18 ms: 20
>>>>>> 2011-06-10 11:10:28.513195 [DEBUG] switch_rtp.c:1607 Starting timer
>>>>>> [soft]
>>>>>> 160 bytes per 20ms
>>>>>> 2011-06-10 11:10:28.514938 [DEBUG] sofia_glue.c:3228 Set 2833 dtmf
>>>>>> send
>>>>>> payload to 101
>>>>>> 2011-06-10 11:10:28.514938 [DEBUG] sofia_glue.c:3233 Set 2833 dtmf
>>>>>> receive
>>>>>> payload to 101
>>>>>> 2011-06-10 11:10:28.514938 [NOTICE] sofia_glue.c:3680 Pre-Answer
>>>>>> sofia/external/03996563750914 at 203.208.207.212!
>>>>>> 2011-06-10 11:10:28.514938 [DEBUG] switch_channel.c:2627
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Callstate Change
>>>>>> RINGING ->
>>>>>> EARLY
>>>>>> 2011-06-10 11:10:28.514938 [DEBUG] switch_ivr_originate.c:3408
>>>>>> Originate
>>>>>> Resulted in Success: [sofia/external/03996563750914 at 203.208.207.212]
>>>>>> 2011-06-10 11:10:28.514938 [DEBUG] mod_commands.c:3205
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change
>>>>>> CS_CONSUME_MEDIA -> CS_EXECUTE
>>>>>> 2011-06-10 11:10:28.514938 [DEBUG] switch_core_session.c:1116 Send
>>>>>> signal
>>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>>> 2011-06-10 11:10:28.515884 [DEBUG] switch_core_state_machine.c:320
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>>> CS_EXECUTE
>>>>>> 2011-06-10 11:10:28.515884 [DEBUG] switch_core_state_machine.c:366
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State EXECUTE
>>>>>> 2011-06-10 11:10:28.515884 [DEBUG] mod_sofia.c:240
>>>>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA EXECUTE
>>>>>> 2011-06-10 11:10:28.515884 [DEBUG] switch_core_state_machine.c:157
>>>>>> sofia/external/03996563750914 at 203.208.207.212 Standard EXECUTE
>>>>>> EXECUTE sofia/external/03996563750914 at 203.208.207.212 park()
>>>>>> 2011-06-10 11:10:28.516887 [DEBUG] switch_core_session.c:954 Send
>>>>>> signal
>>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_ivr_bridge.c:1480
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change
>>>>>> CS_EXECUTE ->
>>>>>> CS_HIBERNATE
>>>>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:1116 Send
>>>>>> signal
>>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_ivr_bridge.c:1482
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change
>>>>>> CS_EXECUTE ->
>>>>>> CS_HIBERNATE
>>>>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:1116 Send
>>>>>> signal
>>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:771 Send
>>>>>> signal
>>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:771 Send
>>>>>> signal
>>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_ivr_play_say.c:1581 done
>>>>>> playing
>>>>>> file
>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_ivr.c:563
>>>>>> sofia/external/03996590031055 at 203.208.207.212 Command Execute
>>>>>> playback(tone_stream://%(2000,4000,440,480);loops=10)
>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:366
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State EXECUTE going
>>>>>> to sleep
>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:320
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>>> CS_HIBERNATE
>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:381
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State HIBERNATE
>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] mod_sofia.c:221
>>>>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA HIBERNATE
>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_ivr_bridge.c:731
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change
>>>>>> CS_HIBERNATE ->
>>>>>> CS_RESET
>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_session.c:1116 Send
>>>>>> signal
>>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:381
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State HIBERNATE going
>>>>>> to
>>>>>> sleep
>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:320
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>>> CS_RESET
>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:362
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State RESET
>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] mod_sofia.c:165
>>>>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA RESET
>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_ivr_bridge.c:716
>>>>>> sofia/external/03996563750914 at 203.208.207.212 CUSTOM RESET
>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:66
>>>>>> sofia/external/03996563750914 at 203.208.207.212 Standard RESET
>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:362
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State RESET going to
>>>>>> sleep
>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_session.c:709 Send
>>>>>> signal
>>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>>> recv 666 bytes from udp/[203.208.207.212]:5060 at 03:10:30.540814:
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>>   SIP/2.0 183 Session Progress
>>>>>>   Via: SIP/2.0/UDP
>>>>>> 10.1.1.46:5080
>>>>>> ;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>>>>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>>>>>   To:
>>>>>> &lt;sip:03996563750914 at 203.208.207.212
>>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
>>>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>>>   CSeq: 13501649 INVITE
>>>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER,
>>>>>> NOTIFY,
>>>>>> SUBSCRIBE, UPDATE
>>>>>>   Content-Type: application/sdp
>>>>>>   Content-Length:   186
>>>>>>
>>>>>>   v=0
>>>>>>   o=- 131082 2 IN IP4 203.208.207.218
>>>>>>   s=session
>>>>>>   c=IN IP4 203.208.207.195
>>>>>>   t=0 0
>>>>>>   m=audio 45002 RTP/AVP 18 101
>>>>>>   a=rtpmap:18 G729/8000
>>>>>>   a=rtpmap:101 telephone-event/8000
>>>>>>   a=sendrecv
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>> 2011-06-10 11:10:30.435309 [INFO] sofia.c:729
>>>>>> sofia/external/03996563750914 at 203.208.207.212 Update Callee ID to
>>>>>> "03996563750914" <03996563750914>
>>>>>> 2011-06-10 11:10:30.435309 [DEBUG] sofia.c:4641 Channel
>>>>>> sofia/external/03996563750914 at 203.208.207.212 skipping state
>>>>>> [proceeding][183]
>>>>>> 2011-06-10 11:10:30.691404 [WARNING] switch_core_session.c:1940 Cannot
>>>>>> execute app 'playback' media required on an outbound channel that does
>>>>>> not
>>>>>> have media established
>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:366
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State EXECUTE going
>>>>>> to sleep
>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>>> CS_HIBERNATE
>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:381
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State HIBERNATE
>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:221
>>>>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA HIBERNATE
>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:731
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change
>>>>>> CS_HIBERNATE ->
>>>>>> CS_RESET
>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_session.c:1116 Send
>>>>>> signal
>>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:381
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State HIBERNATE going
>>>>>> to
>>>>>> sleep
>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>>> CS_RESET
>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:362
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State RESET
>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:165
>>>>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA RESET
>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:716
>>>>>> sofia/external/03996590031055 at 203.208.207.212 CUSTOM RESET
>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:723
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_RESET
>>>>>> ->
>>>>>> CS_SOFT_EXECUTE
>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_session.c:1116 Send
>>>>>> signal
>>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:362
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State RESET going to
>>>>>> sleep
>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>>> CS_SOFT_EXECUTE
>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:372
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State SOFT_EXECUTE
>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:558 SOFIA SOFT_EXECUTE
>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:741
>>>>>> sofia/external/03996590031055 at 203.208.207.212 CUSTOM SOFT_EXECUTE
>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:761
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_RESET
>>>>>> ->
>>>>>> CS_SOFT_EXECUTE
>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_session.c:1116 Send
>>>>>> signal
>>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>>> CS_SOFT_EXECUTE
>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:372
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State SOFT_EXECUTE
>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:558 SOFIA SOFT_EXECUTE
>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:741
>>>>>> sofia/external/03996563750914 at 203.208.207.212 CUSTOM SOFT_EXECUTE
>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:204
>>>>>> sofia/external/03996563750914 at 203.208.207.212 Standard SOFT_EXECUTE
>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:372
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State SOFT_EXECUTE
>>>>>> going to
>>>>>> sleep
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2535
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Callstate Change
>>>>>> EARLY ->
>>>>>> HANGUP
>>>>>> 2011-06-10 11:10:30.711798 [NOTICE] switch_ivr_originate.c:1045 Hangup
>>>>>> sofia/external/03996563750914 at 203.208.207.212 [CS_SOFT_EXECUTE]
>>>>>> [DESTINATION_OUT_OF_ORDER]
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2551 Send signal
>>>>>> sofia/external/03996563750914 at 203.208.207.212 [KILL]
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send
>>>>>> signal
>>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>>> CS_HANGUP
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2535
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change
>>>>>> ACTIVE ->
>>>>>> HANGUP
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State HANGUP
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:457 Channel
>>>>>> sofia/external/03996563750914 at 203.208.207.212 hanging up, cause:
>>>>>> DESTINATION_OUT_OF_ORDER
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:510 Sending CANCEL to
>>>>>> sofia/external/03996563750914 at 203.208.207.212
>>>>>> send 390 bytes to udp/[203.208.207.212]:5060 at 03:10:30.818391:
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>>   CANCEL sip:03996563750914 at 203.208.207.212 SIP/2.0
>>>>>>   Via: SIP/2.0/UDP 202.73.56.46:5080
>>>>>> ;rport;branch=z9hG4bKZ49rj5Qtr8aBN
>>>>>>   Max-Forwards: 70
>>>>>>   From: "" &lt;sip:03996590031055 at 202.73.56.46&gt;;tag=jDSe6ycN62Zar
>>>>>>   To: &lt;sip:03996563750914 at 203.208.207.212&gt;
>>>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>>>   CSeq: 13501649 CANCEL
>>>>>>   Reason: Q.850;cause=27;text="DESTINATION_OUT_OF_ORDER"
>>>>>>   Content-Length: 0
>>>>>>
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>> 2011-06-10 11:10:30.711798 [NOTICE] switch_ivr_bridge.c:772 Hangup
>>>>>> sofia/external/03996590031055 at 203.208.207.212 [CS_SOFT_EXECUTE]
>>>>>> [ORIGINATOR_CANCEL]
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:46
>>>>>> sofia/external/03996563750914 at 203.208.207.212 Standard HANGUP, cause:
>>>>>> DESTINATION_OUT_OF_ORDER
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State HANGUP going to
>>>>>> sleep
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:351
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change
>>>>>> CS_HANGUP ->
>>>>>> CS_REPORTING
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send
>>>>>> signal
>>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>>> CS_REPORTING
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:617
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State REPORTING
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:53
>>>>>> sofia/external/03996563750914 at 203.208.207.212 Standard REPORTING,
>>>>>> cause:
>>>>>> DESTINATION_OUT_OF_ORDER
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:617
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State REPORTING going
>>>>>> to
>>>>>> sleep
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2551 Send signal
>>>>>> sofia/external/03996590031055 at 203.208.207.212 [KILL]
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send
>>>>>> signal
>>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:372
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State SOFT_EXECUTE
>>>>>> going to
>>>>>> sleep
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>>> CS_HANGUP
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:345
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change
>>>>>> CS_REPORTING ->
>>>>>> CS_DESTROY
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send
>>>>>> signal
>>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1288 Session
>>>>>> 40
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Locked, Waiting on
>>>>>> external
>>>>>> entities
>>>>>> 2011-06-10 11:10:30.711798 [NOTICE] switch_core_session.c:1306 Session
>>>>>> 40
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Ended
>>>>>> 2011-06-10 11:10:30.711798 [NOTICE] switch_core_session.c:1308 Close
>>>>>> Channel
>>>>>> sofia/external/03996563750914 at 203.208.207.212 [CS_DESTROY]
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:449
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Callstate Change
>>>>>> HANGUP ->
>>>>>> DOWN
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:452
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>>> CS_DESTROY
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:462
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State DESTROY
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:362
>>>>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA DESTROY
>>>>>> 2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
>>>>>> 0x2aaaac013028 (nil)
>>>>>> 2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
>>>>>> 0x2aaaac013028 (nil)
>>>>>> 2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
>>>>>> 0x2aaaac013088 (nil)
>>>>>> 2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
>>>>>> 0x2aaaac013088 (nil)
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:60
>>>>>> sofia/external/03996563750914 at 203.208.207.212 Standard DESTROY
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:462
>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State DESTROY going
>>>>>> to sleep
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State HANGUP
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:457 Channel
>>>>>> sofia/external/03996590031055 at 203.208.207.212 hanging up, cause:
>>>>>> ORIGINATOR_CANCEL
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:500 Sending BYE to
>>>>>> sofia/external/03996590031055 at 203.208.207.212
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:46
>>>>>> sofia/external/03996590031055 at 203.208.207.212 Standard HANGUP, cause:
>>>>>> ORIGINATOR_CANCEL
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State HANGUP going to
>>>>>> sleep
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:351
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change
>>>>>> CS_HANGUP ->
>>>>>> CS_REPORTING
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send
>>>>>> signal
>>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>>> CS_REPORTING
>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:617
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State REPORTING
>>>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:53
>>>>>> sofia/external/03996590031055 at 203.208.207.212 Standard REPORTING,
>>>>>> cause:
>>>>>> ORIGINATOR_CANCEL
>>>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:617
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State REPORTING going
>>>>>> to
>>>>>> sleep
>>>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:345
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change
>>>>>> CS_REPORTING ->
>>>>>> CS_DESTROY
>>>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_session.c:1116 Send
>>>>>> signal
>>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_session.c:1288 Session
>>>>>> 39
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Locked, Waiting on
>>>>>> external
>>>>>> entities
>>>>>> 2011-06-10 11:10:30.714001 [NOTICE] switch_core_session.c:1306 Session
>>>>>> 39
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Ended
>>>>>> 2011-06-10 11:10:30.714001 [NOTICE] switch_core_session.c:1308 Close
>>>>>> Channel
>>>>>> sofia/external/03996590031055 at 203.208.207.212 [CS_DESTROY]
>>>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:449
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change
>>>>>> HANGUP ->
>>>>>> DOWN
>>>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:452
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>>> CS_DESTROY
>>>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:462
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State DESTROY
>>>>>> 2011-06-10 11:10:30.714001 [DEBUG] mod_sofia.c:362
>>>>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA DESTROY
>>>>>> 2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
>>>>>> 0x2aaab0031060 (nil)
>>>>>> 2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
>>>>>> 0x2aaab0031060 (nil)
>>>>>> 2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
>>>>>> 0x2aaab00310c0 0x2aaab00b20c0
>>>>>> 2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
>>>>>> 0x2aaab00310c0 (nil)
>>>>>> send 662 bytes to udp/[203.208.207.212]:5060 at 03:10:30.820530:
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>>   BYE sip:203.208.207.212:5060 SIP/2.0
>>>>>>   Via: SIP/2.0/UDP 202.73.56.46:5080
>>>>>> ;rport;branch=z9hG4bK0D3Hm08XNH1Xg
>>>>>>   Max-Forwards: 70
>>>>>>   From: "" &lt;sip:03996563750914 at 202.73.56.46&gt;;tag=H4ZN43UH9S9Qc
>>>>>>   To:
>>>>>> &lt;sip:03996590031055 at 203.208.207.212
>>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>>>   CSeq: 13501634 BYE
>>>>>>   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
>>>>>>   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
>>>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>>>> REGISTER, REFER, NOTIFY
>>>>>>   Supported: timer, precondition, path, replaces
>>>>>>   Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL"
>>>>>>   Content-Length: 0
>>>>>>
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>> 2011-06-10 11:10:30.715878 [INFO] mod_com_g729.c:83 ENCODER DESTROY -
>>>>>> 0x2aaab00310c0 0x2aaab00b20c0
>>>>>> 2011-06-10 11:10:30.715878 [DEBUG] switch_core_state_machine.c:60
>>>>>> sofia/external/03996590031055 at 203.208.207.212 Standard DESTROY
>>>>>> 2011-06-10 11:10:30.715878 [DEBUG] switch_core_state_machine.c:462
>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State DESTROY going
>>>>>> to sleep
>>>>>> recv 383 bytes from udp/[203.208.207.212]:5060 at 03:10:30.823302:
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>>   SIP/2.0 200 Ok
>>>>>>   Via: SIP/2.0/UDP
>>>>>> 10.1.1.46:5080
>>>>>> ;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>>>>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>>>>>   To:
>>>>>> &lt;sip:03996563750914 at 203.208.207.212
>>>>>> &gt;;tag=2QGB951HCR30000E1D00000u00000001QXU3LU
>>>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>>>   CSeq: 13501649 CANCEL
>>>>>>   Contact: <sip:203.208.207.212:5060>
>>>>>>   Content-Length:     0
>>>>>>
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>> recv 411 bytes from udp/[203.208.207.212]:5060 at 03:10:30.824765:
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>>   SIP/2.0 487 Request Terminated
>>>>>>   Via: SIP/2.0/UDP
>>>>>> 10.1.1.46:5080
>>>>>> ;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>>>>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>>>>>   To:
>>>>>> &lt;sip:03996563750914 at 203.208.207.212
>>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
>>>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>>>   CSeq: 13501649 INVITE
>>>>>>   Reason: SIP;cause=487;text="Request Terminated"
>>>>>>   Content-Length:     0
>>>>>>
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>> send 371 bytes to udp/[203.208.207.212]:5060 at 03:10:30.824891:
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>>   ACK sip:03996563750914 at 203.208.207.212 SIP/2.0
>>>>>>   Via: SIP/2.0/UDP 202.73.56.46:5080
>>>>>> ;rport;branch=z9hG4bKZ49rj5Qtr8aBN
>>>>>>   Max-Forwards: 70
>>>>>>   From: "" &lt;sip:03996590031055 at 202.73.56.46&gt;;tag=jDSe6ycN62Zar
>>>>>>   To:
>>>>>> &lt;sip:03996563750914 at 203.208.207.212
>>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
>>>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>>>   CSeq: 13501649 ACK
>>>>>>   Content-Length: 0
>>>>>>
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>> recv 380 bytes from udp/[203.208.207.212]:5060 at 03:10:30.826375:
>>>>>>
>>>>>> ------------------------------------------------------------------------
>>>>>>   SIP/2.0 200 Ok
>>>>>>   Via: SIP/2.0/UDP
>>>>>> 10.1.1.46:5080
>>>>>> ;rport=5080;branch=z9hG4bK0D3Hm08XNH1Xg;received=10.1.1.46
>>>>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>>>>   To:
>>>>>> &lt;sip:03996590031055 at 203.208.207.212
>>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>>>   CSeq: 13501634 BYE
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> View this message in context:
>>>>>> http://freeswitch-users.2379917.n2.nabble.com/Leg-A-is-automatically-disconnected-on-Leg-B-orginate-failure-tp6482192p6482192.html
>>>>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Jun 27 19:15:36 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 27 Jun 2011 08:15:36 -0700
Subject: [Freeswitch-users] Please let's protect the wiki from spammers
In-Reply-To: <BANLkTi=cK4CUE6jiYAGs3Rb6iQWzRVRtMQ@mail.gmail.com>
References: <BANLkTinVTUtGPVXCa_3=Lhrpe2esqB2dXw@mail.gmail.com>
	<BANLkTi=cK4CUE6jiYAGs3Rb6iQWzRVRtMQ@mail.gmail.com>
Message-ID: <BANLkTinCS5i4k4nSzwMTSaVqVjEX=0+49A@mail.gmail.com>

I am okay with updating the wiki so that we can do the captcha thing. I need
to work w/ Raymond (intralanman) on some back-end infrastructure stuff
before that happens. We'll discuss this further on the Wednesday conf call
this week.
-MC

On Sun, Jun 26, 2011 at 7:52 AM, Avi Marcus <avi at avimarcus.net> wrote:

> Linode uses .htaccess on the edit pages (only those), much like our
> pastebin, and is probably the easiest thing to add.
> But yes, something needs to be done.. I got sick of deleting spam months
> ago.
>
> A captcha on normal accounts to make edits is probably OK, too, as long as
> they can manually be moved to "human-trusted".
> -Avi
>
>
>
> On Fri, Jun 24, 2011 at 10:36 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> Hi,
>>
>> As most of us may alright know, the FreeSWITCH wiki has been abused
>> and attacked by spammers, as seen here:
>> http://wiki.freeswitch.org/wiki/Special:RecentChanges
>>
>> Some FreeSWITCH users have been deleting and blocking these spammers,
>> but they are increasing their spamming attacks.
>>
>> I went into #mediawiki in freenode and asked them if there is any
>> solution to this and this is what they suggest:
>>
>> 15:18 < bawolff> !spam | diegoviola
>> 15:18 < mw-bot> diegoviola: For information about combating and
>> handling spam in MediaWiki, see
>> <http://www.mediawiki.org/wiki/Manual:Combating_spam> and
>> <http://www.mediawiki.org/wiki/Anti-spam_features>.
>>
>>
>> I've seen most wikis who have implemented this solution and they no
>> longer get any spamming after applying said solution. If somebody here
>> have administration access to http://wiki.freeswitch.org/ I offer my
>> help to apply this solution.
>>
>> Thanks,
>> Diego Viola
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Jun 27 19:25:44 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 27 Jun 2011 08:25:44 -0700
Subject: [Freeswitch-users] Phrase macros in conferences
In-Reply-To: <FE02E1C7-64EA-4E85-AC33-B491AB2FF555@bryansmart.com>
References: <FE02E1C7-64EA-4E85-AC33-B491AB2FF555@bryansmart.com>
Message-ID: <BANLkTikRKyF8eppAtqd6DbQqRH29JLnQ7w@mail.gmail.com>

Phrase macros are not supported inside mod_conference. If you look inside
mod_conference.c you will see that it has its own functions for injecting
audio into the conference and it does not support phrase macros. You can
open up a feature request on Jira. I honestly don't know how easy/difficult
this would be, so you may consider offering a bounty to sweeten the deal.

Curious: what types of phrases are you trying to play into the conference?

-MC

On Sun, Jun 26, 2011 at 4:57 PM, Bryan Smart <bryansmart at bryansmart.com>wrote:

> I'm attempting to use phrase macros in mod_conference, but I'm not having
> success. Perhaps they aren't supported?
>
> I have a working phrase macro. In the dialplan, both
>
> <action application="phrase" data="my_phrase"/>
>
> and
>
> <action application="playback" data="phrase:my_phrase"/>
>
> Work just fine.
>
> The wiki says that, in conference.conf.xml, entries like pin-sound accept,
> for their value, file paths, strings prefixed with "say" for TTS, and
> strings prefixed with "tone_stream". This looks almost the same as what can
> be passed to the playback application in the dialplan. When I try to prefix
> a string with "phrase", it doesn't work.
>
> <param name="pin-sound" value="phrase:my_phrase"/>
>
> The console tells me "unknown file format". The console still tells me this
> if I make up phrase names
>
> <param name="pin-sound" value="phrase:not_a_real_phrase"/>
>
> So it is complaining about the syntax or capability, rather than an
> unrecognized format of an actual file.
>
> If mod_conference doesn't support phrase macros, is this a near-future
> feature? Phrases would seem to be necessary to ocalize announcements and
> play announcements that are stitched together from multiple files.
>
> Bryan
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Jun 27 19:45:36 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 27 Jun 2011 08:45:36 -0700
Subject: [Freeswitch-users] Answer issue on inbound call
In-Reply-To: <BANLkTikcWDbCkTPRow-TMqPN=D3FV+LjWw@mail.gmail.com>
References: <BANLkTin+f=uDMiOouyLWAorPvNTd1Z6xaA@mail.gmail.com>
	<BANLkTim=TOQD1tUvmHjeRuhAR-eKkGYgsw@mail.gmail.com>
	<BANLkTim3pTPV804t_Usk59eG2C56HAyFdA@mail.gmail.com>
	<BANLkTi=Q3qRBCDRARDWp15=Guwyq-jtNLQ@mail.gmail.com>
	<BANLkTinMokGeJU+wVVA0U-oxB97z+c+ZGg@mail.gmail.com>
	<BANLkTinEp1Cu5mB_UHw-Y+mitGsNByR3sQ@mail.gmail.com>
	<BANLkTikcWDbCkTPRow-TMqPN=D3FV+LjWw@mail.gmail.com>
Message-ID: <BANLkTinE1292_p_VZd+D4N6vvEEot8LXRw@mail.gmail.com>

Get a complete, unedited, unabridged console debug log w/ siptrace and put
it on pastebin w/ "FreeSWITCH Log" for the syntax highlighting. Use "sofia
global siptrace on" to make sure you can all SIP traffic.

-MC

On Mon, Jun 27, 2011 at 6:05 AM, Max Alex <max.asterisk at gmail.com> wrote:

> Hi,
> Thanks for reply,
> I have tried the same way and reloaded freeswitch, but still it is answered
> on first ring of the call.
> When it is ringing the call on 1001 and the same time it is answered on
> cell phone, so something is done when it is ringing on 1001.
>
> Here is the dialplan for the same
>
>     <extension name="Local_Extension">
>       <condition field="destination_number" expression="^(10[01][0-9])$">
>         <action application="set" data="dialed_extension=$1"/>
>         <action application="export" data="dialed_extension=$1"/>
>         <action application="bind_meta_app" data="2 b s
> record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>         <action application="bind_meta_app" data="3 b s
> execute_extension::cf XML features"/>-->
>         <action application="set" data="ringback=${us-ring}"/>
>         <action application="set" data="transfer_ringback=$${hold_music}"/>
>         <action application="set" data="call_timeout=30"/>
>           <action application="set" data="hangup_after_bridge=true"/>
>          <action application="set" data="continue_on_fail=true"/>
>         <action application="hash"
> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
>         <action application="hash"
> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>         <action application="set"
> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
> var callgroup)}"/>
>         <action application="hash"
> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>
>        <action application="bridge"
> data="{ignore_early_media=true}user/${dialed_extension}@${domain_name}"/>
>        <action application="sleep" data="1000"/>
>         <action application="voicemail" data="default ${domain_name}
> ${dialed_extension}"/>
>       </condition>
>     </extension>
>
>
> Please help me for this issue.
>
>
> Thanks,
> Max Alex
> Voip Developer
>
>
>
> On Fri, Jun 24, 2011 at 11:38 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Are you using the default dialplan? I think you might just need to ignore
>> early media on your bridge app. If you are using the default.xml file then
>> locate "Local_Extension" and the bridge line:
>>
>>     <action application="bridge" data="user/${dialed_extension}@
>> ${domain_name}"/>
>>
>> Change it to this, then try again:
>>
>>     <action application="bridge"
>> data="{ignore_early_media=true}user/${dialed_extension}@${domain_name}"/>
>>
>> If I understand correctly, the "symptom" you are experiencing is the
>> normal operation of the bridge app (and it's cousin, the originate API
>> command). When the b-leg sends back media, including ringing, the bridge (or
>> the originate) will be considered "successful," and in the case of bridge,
>> the b-leg's audio (the early media) will be connected to the a-leg. If you
>> set ignore_early_media=true then you are explicitly telling the bridge app
>> that you only want to connect the b-leg to the a-leg if the b-leg actually
>> answers.
>>
>> I hope that made sense...
>>
>> -MC
>>
>>
>>
>> On Thu, Jun 23, 2011 at 9:32 PM, Max Alex <max.asterisk at gmail.com> wrote:
>>
>>> Hi,
>>> Thanks for reply.
>>> Current scenario is below.
>>>
>>> PSTN call -> sangoma device -> freeswitch incoming context -> routed to
>>> 1001 -> ringing (Answered on cellphone)
>>> Here when it is routed to 1001 the call it is started ringing, but on
>>> phone that call is answered and hearding sound of ringing.
>>>
>>> Required flow:
>>> PSTN call -> sangoma device -> freeswitch incoming context -> routed to
>>> 1001 -> ringing (Ringing on cellphone)
>>>
>>> I hope it is understable, the call should not be answered until 1001
>>> answer it, right not when it is started ring it is answered on cell phone.
>>> that should not be happen as it is not answered yet.
>>>
>>> Waiting for your reply.
>>>
>>>
>>> Thanks,
>>> Max Alex
>>> Voip Developer
>>>
>>>
>>>
>>> On Fri, Jun 24, 2011 at 5:48 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> I'm not sure I understand the problem. What is happening vs. what you
>>>> believe should be happening?
>>>> -MC
>>>>
>>>>
>>>> On Thu, Jun 23, 2011 at 3:31 AM, Max Alex <max.asterisk at gmail.com>wrote:
>>>>
>>>>> Hi,
>>>>> Thanks for your reply.
>>>>> Here is my configuration and log
>>>>> http://pastebin.freeswitch.org/16571
>>>>>
>>>>> I am using A200 analog sangoma device with freeswitch, it is working
>>>>> fine but when it is routing call to 1001 then it is answered.
>>>>> Please provider your suggestions.
>>>>>
>>>>> Thanks,
>>>>> Max Alex
>>>>> Voip Developer
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Wed, Jun 22, 2011 at 8:51 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>
>>>>>> I thought the A200 was an analog card? Maybe I have my numbers mixed
>>>>>> up...
>>>>>>
>>>>>> Go ahead and collect a debug log of this call. It might help to have
>>>>>> your configs posted as well. Use pastebin.freeswitch.org. See this
>>>>>> wiki article for tips on how to collect information:
>>>>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>>>>>
>>>>>> -MC
>>>>>>
>>>>>> On Wed, Jun 22, 2011 at 3:23 AM, Max Alex <max.asterisk at gmail.com>wrote:
>>>>>>
>>>>>>> Hi,
>>>>>>> I have installed freeswitch latest version with sangoma card A200 as
>>>>>>> well,
>>>>>>> I have installed and configured freetdm module with wanpipe drivers
>>>>>>> for freeswitch,
>>>>>>> We are properly receiving the inbound calls in public context and
>>>>>>> then we are routing that call to 1001 extension,
>>>>>>> it is properly routing to 1001 as well, but we have one issue while
>>>>>>> routing on 1001.
>>>>>>>
>>>>>>> Here is the issue description.
>>>>>>> I am calling from my cell phone to that DID number of pri line, and
>>>>>>> then it will start ringing on 1001 extension,
>>>>>>> When 1001 extension start ringing the call is answered on my cell
>>>>>>> phone,
>>>>>>> it is something like freeswitch preanswer or autoanswer the call, how
>>>>>>> can i stop this answer call when it is ringing on 1001 extension,
>>>>>>> Waiting for good reply.
>>>>>>>
>>>>>>> Thanks,
>>>>>>> Max Alex
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From wes-fs at 499x.com  Mon Jun 27 20:20:55 2011
From: wes-fs at 499x.com (Wes)
Date: Mon, 27 Jun 2011 11:20:55 -0500
Subject: [Freeswitch-users] getting rid of hanging sessions?
In-Reply-To: <BANLkTinOLY6dv1tB7sZJSKQC4WMiRmEejA@mail.gmail.com>
References: <4E0503A3.5040909@499x.com>
	<BANLkTinOLY6dv1tB7sZJSKQC4WMiRmEejA@mail.gmail.com>
Message-ID: <4E08ADE7.5030109@499x.com>

Ok, I think they're getting stuck if I hang up while recording the call, 
line 54 in the pastebin source.  When I hang up, I see the message 
logged in line 55, but then I think it enters the "while (listen)" loop, 
and never gets out.  Because if I uncomment line 58, it scrolls 
endlessly in the log saying: "flushing digits 2"  "flushing digits 2"  
"flushing digits 2" ...

script
http://pastebin.freeswitch.org/16601

log:
http://pastebin.freeswitch.org/16602

so is the trick to check something else in the while loop like: while 
(listen and not hung up) ?


On 6/24/2011 6:15 PM, Michael Collins wrote:
> The key is in knowing why those are stuck. Do a "show channels" when 
> you see these so-called phantom sessions, then put it on pastebin so 
> we can take a look.
> -MC
>
> On Fri, Jun 24, 2011 at 2:37 PM, Wes <wes-fs at 499x.com 
> <mailto:wes-fs at 499x.com>> wrote:
>
>     in my testing, I'm getting some sessions that are, hung, I guess.
>
>     Status shows:
>
>     UP 0 years, 0 days, 2 hours, 7 minutes, 37 seconds, 752 milliseconds,
>     720 microseconds
>     23 session(s) since startup
>     5 session(s) 0/30
>     1000 session(s) max
>     min idle cpu 0.00/0.00
>
>
>     if I do a shutdown, it waits on them for a while, then eventually
>     shuts
>     down.
>
>     Is there any way to "clean" these up periodically?  Or, what am I
>     doing
>     to leave them out there?
>
>     thanks!
>
>     _______________________________________________
>     Join us at ClueCon 2011, Aug 9-11, Chicago
>     http://www.cluecon.com 877-7-4ACLUE
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From elijah at crankenstein.com  Mon Jun 27 20:27:38 2011
From: elijah at crankenstein.com (elijah)
Date: Mon, 27 Jun 2011 09:27:38 -0700
Subject: [Freeswitch-users] second member to dynamic conference hears
 only hold music
In-Reply-To: <BANLkTinQL+jaHU0f87MPHNLhr51K4zB2-Q@mail.gmail.com>
References: <BANLkTikxRYXfOELxVNV9gD6Wx=Y9SN4=vA@mail.gmail.com>
	<BANLkTi=9TWsFmhnqwpom6CDN8U_v5fQYDA@mail.gmail.com>
	<BANLkTi=+nT1LT_8pB_7yy_rMf37TFixwcA@mail.gmail.com>
	<BANLkTinQL+jaHU0f87MPHNLhr51K4zB2-Q@mail.gmail.com>
Message-ID: <BANLkTik4CeE3KK267WZBQ82K5KAB4_41tQ@mail.gmail.com>

done - http://jira.freeswitch.org/<http://jira.freeswitch.org/browse/FS-3374>
browse/FS-3374 <http://jira.freeswitch.org/browse/FS-3374>


On Thu, Jun 23, 2011 at 8:42 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> this really should be in jira.
> please open one http://jira.freeswitch.org
>
> and post the url here
>
> do the following
>
> 1) update right now to the latest GIT as of the time you read this.
> 2) turn on debugging and sip trace with these commands:
> >console loglevel debug
> >sofia global siptrace on
> 3) reproduce your issue and capture the console log
> 4) attach it to your jira report
>
> Then:
>
> try version 9df8169d1f3458e0b565a64922a1390ebf324703
>
>
>
>
> On Wed, Jun 22, 2011 at 8:00 PM, elijah <elijah at crankenstein.com> wrote:
> > That didn't help. Maybe I've screwed this up by running 'git reset
> --hard'?
> > I've confirmed this version works for me: FreeSWITCH Version 1.0.head
> > (git-5923f71 2011-06-01 22-36-19 -0500)
> > and this does not: FreeSWITCH Version 1.0.head (git-288455c 2011-06-22
> > 17-05-53 -0400)
> > On Wed, Jun 22, 2011 at 5:30 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> could you comment line 1696 of switch_channel.c and
> >> make install_core
> >> then restart and retest?
> >>
> >> That is the only one I can think of.
> >>
> >> On Wed, Jun 22, 2011 at 5:53 PM, elijah <elijah at crankenstein.com>
> wrote:
> >> > This setup was working for me until updating against git yesterday.
> Via
> >> > a
> >> > bind statement on the b-leg of inbound calls, a user is able to pull
> >> > both
> >> > legs into a conference bridge. The problem now is that the a-leg hears
> >> > only
> >> > hold music. I'm having this and another problem with my dialplan after
> >> > updating, w/o changing my own configuration - could there be updates
> to
> >> > source that would be affecting me? Here's the setup:
> >> >     <extension name="telifi_sales_queue">
> >> >       <condition field="destination_number"
> >> > expression="^telifi_sales_queue$">
> >> >         <action application="bridge_export"
> >> > data="telifi_call_id=${telifi_call_id}"/>
> >> >         <action application="set" data="bind_meta_key=#"/>
> >> >         <action application="bind_meta_app" data="1 b s
> >> > execute_extension::dx1 XML callsdirect"/>
> >> >         <action application="bind_meta_app" data="2 b s
> >> > execute_extension::dx2 XML callsdirect"/>
> >> >         <action application="callcenter"
> >> > data="${telifi_queue_name}@live001.voice.telifi.com"/>
> >> >       </condition>
> >> >     </extension>
> >> >     <extension name="bind_receiver_2">
> >> >       <condition field="destination_number" expression="^dx2$">
> >> >         <action application="answer"/>
> >> >         <action application="transfer" data="-both dynamic_conference
> >> > XML
> >> > callsdirect"/>
> >> >       </condition>
> >> >     </extension>
> >> >     <extension name="dynamic_conference">
> >> >       <condition field="destination_number"
> >> > expression="^dynamic_conference"/>
> >> >       <condition field="rdnis" expression="^(dx2)$">
> >> >         <action application="answer"/>
> >> >         <action application="bind_digit_action"
> >> >
> >> >
> data="moderator,*1,exec:execute_extension,ASK_FOR_NUMBER__${telifi_call_id}
> >> > XML callsdirect"/>
> >> >         <action application="bind_digit_action"
> >> >
> >> >
> data="moderator,*2,exec:execute_extension,CANCEL_LAST_CALL__${telifi_call_id}
> >> > XML callsdirect"/>
> >> >         <action application="digit_action_set_realm"
> data="moderator"/>
> >> >         <action application="conference"
> >> > data="${telifi_call_id}@simple"/>
> >> >         <anti-action application="answer"/>
> >> >         <anti-action application="conference"
> >> > data="${telifi_call_id}@simple"/>
> >> >       </condition>
> >> >     </extension>
> >> >
> >> > _______________________________________________
> >> > Join us at ClueCon 2011, Aug 9-11, Chicago
> >> > http://www.cluecon.com 877-7-4ACLUE
> >> >
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org
> >> googletalk:conf+888 at conference.freeswitch.org
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> Join us at ClueCon 2011, Aug 9-11, Chicago
> >> http://www.cluecon.com 877-7-4ACLUE
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Mon Jun 27 21:27:23 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 27 Jun 2011 18:27:23 +0100
Subject: [Freeswitch-users] getting rid of hanging sessions?
In-Reply-To: <4E08ADE7.5030109@499x.com>
References: <4E0503A3.5040909@499x.com>
	<BANLkTinOLY6dv1tB7sZJSKQC4WMiRmEejA@mail.gmail.com>
	<4E08ADE7.5030109@499x.com>
Message-ID: <BANLkTinzP3ZGC6XeM2QFFcjqONR3ej1jxg@mail.gmail.com>

You need to check session:ready in every loop

-Steve


On 27 June 2011 17:20, Wes <wes-fs at 499x.com> wrote:

> **
> Ok, I think they're getting stuck if I hang up while recording the call,
> line 54 in the pastebin source.  When I hang up, I see the message logged in
> line 55, but then I think it enters the "while (listen)" loop, and never
> gets out.  Because if I uncomment line 58, it scrolls endlessly in the log
> saying: "flushing digits 2"  "flushing digits 2"  "flushing digits 2" ...
>
> script
> http://pastebin.freeswitch.org/16601
>
> log:
> http://pastebin.freeswitch.org/16602
>
> so is the trick to check something else in the while loop like: while
> (listen and not hung up) ?
>
>
>
> On 6/24/2011 6:15 PM, Michael Collins wrote:
>
> The key is in knowing why those are stuck. Do a "show channels" when you
> see these so-called phantom sessions, then put it on pastebin so we can take
> a look.
> -MC
>
> On Fri, Jun 24, 2011 at 2:37 PM, Wes <wes-fs at 499x.com> wrote:
>
>> in my testing, I'm getting some sessions that are, hung, I guess.
>>
>> Status shows:
>>
>> UP 0 years, 0 days, 2 hours, 7 minutes, 37 seconds, 752 milliseconds,
>> 720 microseconds
>> 23 session(s) since startup
>> 5 session(s) 0/30
>> 1000 session(s) max
>> min idle cpu 0.00/0.00
>>
>>
>> if I do a shutdown, it waits on them for a while, then eventually shuts
>> down.
>>
>> Is there any way to "clean" these up periodically?  Or, what am I doing
>> to leave them out there?
>>
>> thanks!
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From wes-fs at 499x.com  Mon Jun 27 21:30:39 2011
From: wes-fs at 499x.com (Wes)
Date: Mon, 27 Jun 2011 12:30:39 -0500
Subject: [Freeswitch-users] How to deal with user pressing extra keys?
 flush_dtmf? session::flushDigits?
In-Reply-To: <BANLkTikvF7pxRqZw+_Dh=EWfmSKRqB-CwA@mail.gmail.com>
References: <4E04EAA7.2050305@499x.com>	<BANLkTi=aaaCs=nXzDU9R_9C1-OLZMFZhiA@mail.gmail.com>	<4E04FC5D.4010900@499x.com>
	<BANLkTikvF7pxRqZw+_Dh=EWfmSKRqB-CwA@mail.gmail.com>
Message-ID: <4E08BE3F.30202@499x.com>

Yes, that is your example modified a bit!

I did not pastebin the latest version, but I did have it in there.  
anyway, it seems to be working better now when I tested it again.

My intention is to detect a human, so that if I get a voicemail system, 
it won't end up recording.

Is there a difference between:

session:execute('flush_dtmf');
and
session:flushDigits()

?


On 6/24/2011 6:13 PM, Michael Collins wrote:
> Hey, that Lua code looks awfully familiar! (I wrote some of that on a 
> snowy day in hotel room in Wisconsin...)
>
> Okay, I think I see what you're trying to do. You just want to have 
> the person press 1 to continue. Do you want something different to 
> happen if the person presses a key other than 1? Just checking.
>
> Also, I don't see any flushDigits in your script. Try adding this 
> before your PAGD:
> session:execute('flush_dtmf');
>
> Try again, report back, etc.
> -MC
>
> On Fri, Jun 24, 2011 at 2:06 PM, Wes <wes-fs at 499x.com 
> <mailto:wes-fs at 499x.com>> wrote:
>
>     I put flushDigits right before playAndGetDigits, but it is still
>     queueing up keypresses... if I type more than one key, weird things
>     start happening in my loop.  (the one that prompts for 1 - play back
>     recording, 2=submit recording, 3 = rerecord)... sometimes I get the
>     message "invalid response" followed a few seconds by "your
>     recording has
>     been submitted", and I've not typed anything in between.
>
>     incidentally, if I press a digit while playAndGetDigits is still
>     speaking, it takes a few seconds for it to stop talking and act on the
>     keypress... ... it just doesn't seem very responsive.... which is why
>     I'm testing extra keypresses... because if users don't immediately
>     get a
>     response, they tend to press the key again.
>
>     script here:
>     http://pastebin.freeswitch.org/16584
>
>
>
>     On 6/24/2011 3:17 PM, Anthony Minessale wrote:
>     > session:flushDigits() right before you collect DTMF and you will
>     never
>     > get any old dtmf
>     >
>     >
>     > On Fri, Jun 24, 2011 at 2:51 PM, Wes<wes-fs at 499x.com
>     <mailto:wes-fs at 499x.com>>  wrote:
>     >> I'm writing an ivr script in lua, and using playAndGetDigits in
>     a couple
>     >> different places.  Both places just take a single digit to
>     continue on.
>     >> If the user presses 2 digits, it seems as if the second digit
>     is being
>     >> queued and it's being processed in the next call to
>     playAndGetDigits.
>     >> Often the extra digit is invalid, so when it tells the user to
>     press a
>     >> certain key, it follows immediately with the message "invalid
>     key", due
>     >> to the extra key being in the "queue"...
>     >>
>     >> is this anything I can protect against?   I found a flush_dtmf,
>     but it
>     >> is not very documented:
>     >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_flush_dtmf
>     >>
>     >> and I also found this:
>     >> http://wiki.freeswitch.org/wiki/Session_flushDigits
>     >>
>     >> but calling it before and after playAndGetDigits didn't *seem* to
>     >> help... should it?
>     >>
>     >> script here:
>     >> http://pastebin.freeswitch.org/16584
>     >>
>     >>
>     >>
>     >>
>     >> _______________________________________________
>     >> Join us at ClueCon 2011, Aug 9-11, Chicago
>     >> http://www.cluecon.com 877-7-4ACLUE
>     >>
>     >> FreeSWITCH-users mailing list
>     >> FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >> http://www.freeswitch.org
>     >>
>     >
>     >
>
>     _______________________________________________
>     Join us at ClueCon 2011, Aug 9-11, Chicago
>     http://www.cluecon.com 877-7-4ACLUE
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From kheimerl at cs.berkeley.edu  Mon Jun 27 22:36:56 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Mon, 27 Jun 2011 11:36:56 -0700
Subject: [Freeswitch-users] Reject SIP registrations
In-Reply-To: <BANLkTimq4YjhjX0vgCt3fsmBcAZRC1OZ-g@mail.gmail.com>
References: <BANLkTikU6fRYwmUTxzsSHyWs8raukLx4jw@mail.gmail.com>
	<BANLkTimq4YjhjX0vgCt3fsmBcAZRC1OZ-g@mail.gmail.com>
Message-ID: <BANLkTikQoYWjH8Nsp20Lg4wsy2O0A3L=4Q@mail.gmail.com>

That would explain why removing them didn't do anything!

Thanks.

On Mon, Jun 27, 2011 at 6:25 AM, Steven Ayre <steveayre at gmail.com> wrote:
> Just so you know...
>
> ? ?? <param name="accept-blind-reg" value="true"/>
> ? ? ?<param name="accept-blind-auth" value="true"/>
>
> These will have no effect in the user directory. They only apply to SIP
> profiles.
>
> -Steve
>
>
>
> On 27 June 2011 02:23, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
>>
>> Hello FS Users!
>>
>> I'm trying to create the following setup. When a user registers, if
>> they register on a known account (lets say X), they do not need a
>> password. X's registration is immediately OK'd, and everything is
>> great. I've gotten that working using the ACL. The IP address of our
>> SIP clients are added through cidr and the clients do not need to give
>> passwords.
>>
>> However, for some reason, if another account that does not exist in
>> the directory (let's say Y) registers, FS returns with a 200 OK,
>> instead of rejecting Y. I'm trying to figure out why this is the case,
>> and how to remedy that fact.
>>
>> I have the following line in my internal.xml file, which I had assumed
>> would force this function:
>> ? <!-- Force the user and auth-user to match. -->
>> ? <param name="inbound-reg-force-matching-username" value="true"/>
>>
>> However, it does not work. In my directory, each individual account as
>> the following lines:
>> ?<user id="1303">
>> ? ?<params>
>> ? ? ?<param name="accept-blind-reg" value="true"/>
>> ? ? ?<param name="accept-blind-auth" value="true"/>
>> ? ? ?<param name="vm-password" value="1000"/>
>> ? ?</params>
>>
>> Though I've found that removing it (from all users in the directory)
>> doesn't help.
>>
>> I'm primarily concerned with the line in internal.xml; it seems
>> possible that the fact that we do not have an auth-user (because we do
>> not require auth) means that this won't work. However, I have yet to
>> test that hypothesis. The ACL has been the most confusing aspect of
>> this installation, with a lot of undocumented aspects, and I get the
>> nagging feeling this is another. I could very well be wrong though.
>>
>> Thanks for any direction.
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From wes-fs at 499x.com  Mon Jun 27 22:51:00 2011
From: wes-fs at 499x.com (Wes)
Date: Mon, 27 Jun 2011 13:51:00 -0500
Subject: [Freeswitch-users] getting rid of hanging sessions?
In-Reply-To: <BANLkTinzP3ZGC6XeM2QFFcjqONR3ej1jxg@mail.gmail.com>
References: <4E0503A3.5040909@499x.com>	<BANLkTinOLY6dv1tB7sZJSKQC4WMiRmEejA@mail.gmail.com>	<4E08ADE7.5030109@499x.com>
	<BANLkTinzP3ZGC6XeM2QFFcjqONR3ej1jxg@mail.gmail.com>
Message-ID: <4E08D114.40705@499x.com>

thank you, works great now!

On 6/27/2011 12:27 PM, Steven Ayre wrote:
> You need to check session:ready in every loop
>
> -Steve
>
>
> On 27 June 2011 17:20, Wes <wes-fs at 499x.com <mailto:wes-fs at 499x.com>> 
> wrote:
>
>     Ok, I think they're getting stuck if I hang up while recording the
>     call, line 54 in the pastebin source.  When I hang up, I see the
>     message logged in line 55, but then I think it enters the "while
>     (listen)" loop, and never gets out.  Because if I uncomment line
>     58, it scrolls endlessly in the log saying: "flushing digits 2" 
>     "flushing digits 2"  "flushing digits 2" ...
>
>     script
>     http://pastebin.freeswitch.org/16601
>
>     log:
>     http://pastebin.freeswitch.org/16602
>
>     so is the trick to check something else in the while loop like:
>     while (listen and not hung up) ?
>
>
>
>     On 6/24/2011 6:15 PM, Michael Collins wrote:
>>     The key is in knowing why those are stuck. Do a "show channels"
>>     when you see these so-called phantom sessions, then put it on
>>     pastebin so we can take a look.
>>     -MC
>>
>>     On Fri, Jun 24, 2011 at 2:37 PM, Wes <wes-fs at 499x.com
>>     <mailto:wes-fs at 499x.com>> wrote:
>>
>>         in my testing, I'm getting some sessions that are, hung, I guess.
>>
>>         Status shows:
>>
>>         UP 0 years, 0 days, 2 hours, 7 minutes, 37 seconds, 752
>>         milliseconds,
>>         720 microseconds
>>         23 session(s) since startup
>>         5 session(s) 0/30
>>         1000 session(s) max
>>         min idle cpu 0.00/0.00
>>
>>
>>         if I do a shutdown, it waits on them for a while, then
>>         eventually shuts
>>         down.
>>
>>         Is there any way to "clean" these up periodically?  Or, what
>>         am I doing
>>         to leave them out there?
>>
>>         thanks!
>>
>>         _______________________________________________
>>         Join us at ClueCon 2011, Aug 9-11, Chicago
>>         http://www.cluecon.com 877-7-4ACLUE
>>
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>     _______________________________________________
>>     Join us at ClueCon 2011, Aug 9-11, Chicago
>>     http://www.cluecon.com  877-7-4ACLUE
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>     _______________________________________________
>     Join us at ClueCon 2011, Aug 9-11, Chicago
>     http://www.cluecon.com 877-7-4ACLUE
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From bryansmart at bryansmart.com  Mon Jun 27 23:05:06 2011
From: bryansmart at bryansmart.com (Bryan Smart)
Date: Mon, 27 Jun 2011 15:05:06 -0400
Subject: [Freeswitch-users] Phrase macros in conferences
In-Reply-To: <BANLkTikRKyF8eppAtqd6DbQqRH29JLnQ7w@mail.gmail.com>
References: <FE02E1C7-64EA-4E85-AC33-B491AB2FF555@bryansmart.com>
	<BANLkTikRKyF8eppAtqd6DbQqRH29JLnQ7w@mail.gmail.com>
Message-ID: <93AE9C49-C6F6-44E6-BC7E-5A31AC659C4C@bryansmart.com>

Thanks, Michael. At least I know that it is a limitation, and not me overlooking something. :) I have 3 or 4 years on Asterisk, but only months on Freeswitch, so I'm always concerned that I haven't dug enough before posting requests.

For phrases inside conferences, I imagine 3 cases.

First are dynamic announcements, such as those that announce when someone joins or leaves a conference. It is helpful if the joining or leaving party can be identified. I imagined combining TTS for the name, along with a pre-recorded fragment like conference/conf-has_joined.wav. The result would be like "John has entered the conference." Right now, I must use entirely TTS for such announcements. I can't use "!" to concatenate files, and I can't defer to a phrase macro to string TTS and recordings together.

Second applies to localization. In a dynamic announcement, it might not be appropriate for the noun to be spoken first in languages other than English. Right now, the conference's prompt language must be customized by creating a different conference profile that specifies the correct sound base for the language. That could mean that the number of profiles in use has to be multiplied by the number of languages that should be supported. I know that the profile can omit the sound base, and the sound base of the first person to join will be used, but, particularly in a situation that concatenates TTS and audio, that can be worse. If the first person to join automatically selects sound files for a language, but the concatenation order is different for that language, then any rules in that profile that concatenate TTS and recordings to render announcements could result in strange outcomes. As far as I've been able to discover, the phrase system is the appropriate Freeswitch way to localize announcements, so it would probably be best for Conference to work that way also, rather than depending on per-case localization being handled with multiple profiles. If phrases were supported, then a single conference profile could play both static and dynamic prompts, and could automatically adjust to any language where phrase macros are present on the system.

Last are cases when multiple types of notification are desired. For example, notifications might use a combination of cues and speech. An audio cue serves a similar function to a visual icon: attracts attention and conveys general meaning. In a situation like a conference, there might be soft audio cues for joining, leaving, when a new person has the floor, etc. When one of those events takes place, the cue plays first to attract attention and set the listener's expectations regarding the type of spoken information that will follow. Next, an announcement is heard with details about the event. "<cue> John has left  the conference."






On Jun 27, 2011, at 11:25 AM, Michael Collins wrote:

Phrase macros are not supported inside mod_conference. If you look inside mod_conference.c you will see that it has its own functions for injecting audio into the conference and it does not support phrase macros. You can open up a feature request on Jira. I honestly don't know how easy/difficult this would be, so you may consider offering a bounty to sweeten the deal.

Curious: what types of phrases are you trying to play into the conference?

-MC

On Sun, Jun 26, 2011 at 4:57 PM, Bryan Smart <bryansmart at bryansmart.com<mailto:bryansmart at bryansmart.com>> wrote:
I'm attempting to use phrase macros in mod_conference, but I'm not having success. Perhaps they aren't supported?

I have a working phrase macro. In the dialplan, both

<action application="phrase" data="my_phrase"/>

and

<action application="playback" data="phrase:my_phrase"/>

Work just fine.

The wiki says that, in conference.conf.xml, entries like pin-sound accept, for their value, file paths, strings prefixed with "say" for TTS, and strings prefixed with "tone_stream". This looks almost the same as what can be passed to the playback application in the dialplan. When I try to prefix a string with "phrase", it doesn't work.

<param name="pin-sound" value="phrase:my_phrase"/>

The console tells me "unknown file format". The console still tells me this if I make up phrase names

<param name="pin-sound" value="phrase:not_a_real_phrase"/>

So it is complaining about the syntax or capability, rather than an unrecognized format of an actual file.

If mod_conference doesn't support phrase macros, is this a near-future feature? Phrases would seem to be necessary to ocalize announcements and play announcements that are stitched together from multiple files.

Bryan


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

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


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

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FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
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From brad at tech21.com  Mon Jun 27 23:37:52 2011
From: brad at tech21.com (Brad Mina)
Date: Mon, 27 Jun 2011 12:37:52 -0700
Subject: [Freeswitch-users] Phrase macros in conferences
In-Reply-To: <93AE9C49-C6F6-44E6-BC7E-5A31AC659C4C@bryansmart.com>
References: <FE02E1C7-64EA-4E85-AC33-B491AB2FF555@bryansmart.com>
	<BANLkTikRKyF8eppAtqd6DbQqRH29JLnQ7w@mail.gmail.com>
	<93AE9C49-C6F6-44E6-BC7E-5A31AC659C4C@bryansmart.com>
Message-ID: <9CE9932A-AD27-4183-9C2E-75148116A23B@tech21.com>

You could have users prompted to record their name to get around the  
first issue and play it back before "... Has joined the conference" is  
played back.

Sent from my iPhone

On Jun 27, 2011, at 12:05 PM, Bryan Smart <bryansmart at bryansmart.com>  
wrote:

> Thanks, Michael. At least I know that it is a limitation, and not me  
> overlooking something. :) I have 3 or 4 years on Asterisk, but only  
> months on Freeswitch, so I'm always concerned that I haven't dug  
> enough before posting requests.
>
> For phrases inside conferences, I imagine 3 cases.
>
> First are dynamic announcements, such as those that announce when  
> someone joins or leaves a conference. It is helpful if the joining  
> or leaving party can be identified. I imagined combining TTS for the  
> name, along with a pre-recorded fragment like conference/conf- 
> has_joined.wav. The result would be like "John has entered the  
> conference." Right now, I must use entirely TTS for such  
> announcements. I can't use "!" to concatenate files, and I can't  
> defer to a phrase macro to string TTS and recordings together.
>
> Second applies to localization. In a dynamic announcement, it might  
> not be appropriate for the noun to be spoken first in languages  
> other than English. Right now, the conference's prompt language must  
> be customized by creating a different conference profile that  
> specifies the correct sound base for the language. That could mean  
> that the number of profiles in use has to be multiplied by the  
> number of languages that should be supported. I know that the  
> profile can omit the sound base, and the sound base of the first  
> person to join will be used, but, particularly in a situation that  
> concatenates TTS and audio, that can be worse. If the first person  
> to join automatically selects sound files for a language, but the  
> concatenation order is different for that language, then any rules  
> in that profile that concatenate TTS and recordings to render  
> announcements could result in strange outcomes. As far as I've been  
> able to discover, the phrase system is the appropriate Freeswitch  
> way to localize announcements, so it would probably be best for  
> Conference to work that way also, rather than depending on per-case  
> localization being handled with multiple profiles. If phrases were  
> supported, then a single conference profile could play both static  
> and dynamic prompts, and could automatically adjust to any language  
> where phrase macros are present on the system.
>
> Last are cases when multiple types of notification are desired. For  
> example, notifications might use a combination of cues and speech.  
> An audio cue serves a similar function to a visual icon: attracts  
> attention and conveys general meaning. In a situation like a  
> conference, there might be soft audio cues for joining, leaving,  
> when a new person has the floor, etc. When one of those events takes  
> place, the cue plays first to attract attention and set the  
> listener's expectations regarding the type of spoken information  
> that will follow. Next, an announcement is heard with details about  
> the event. "<cue> John has left  the conference."
>
>
>
>
>
>
> On Jun 27, 2011, at 11:25 AM, Michael Collins wrote:
>
>> Phrase macros are not supported inside mod_conference. If you look  
>> inside mod_conference.c you will see that it has its own functions  
>> for injecting audio into the conference and it does not support  
>> phrase macros. You can open up a feature request on Jira. I  
>> honestly don't know how easy/difficult this would be, so you may  
>> consider offering a bounty to sweeten the deal.
>>
>> Curious: what types of phrases are you trying to play into the  
>> conference?
>>
>> -MC
>>
>> On Sun, Jun 26, 2011 at 4:57 PM, Bryan Smart <bryansmart at bryansmart.com 
>> > wrote:
>> I'm attempting to use phrase macros in mod_conference, but I'm not  
>> having success. Perhaps they aren't supported?
>>
>> I have a working phrase macro. In the dialplan, both
>>
>> <action application="phrase" data="my_phrase"/>
>>
>> and
>>
>> <action application="playback" data="phrase:my_phrase"/>
>>
>> Work just fine.
>>
>> The wiki says that, in conference.conf.xml, entries like pin-sound  
>> accept, for their value, file paths, strings prefixed with "say"  
>> for TTS, and strings prefixed with "tone_stream". This looks almost  
>> the same as what can be passed to the playback application in the  
>> dialplan. When I try to prefix a string with "phrase", it doesn't  
>> work.
>>
>> <param name="pin-sound" value="phrase:my_phrase"/>
>>
>> The console tells me "unknown file format". The console still tells  
>> me this if I make up phrase names
>>
>> <param name="pin-sound" value="phrase:not_a_real_phrase"/>
>>
>> So it is complaining about the syntax or capability, rather than an  
>> unrecognized format of an actual file.
>>
>> If mod_conference doesn't support phrase macros, is this a near- 
>> future feature? Phrases would seem to be necessary to ocalize  
>> announcements and play announcements that are stitched together  
>> from multiple files.
>>
>> Bryan
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
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From wes-fs at 499x.com  Mon Jun 27 23:43:38 2011
From: wes-fs at 499x.com (Wes)
Date: Mon, 27 Jun 2011 14:43:38 -0500
Subject: [Freeswitch-users] .wav vs .gsm file sizes for recording calls.
In-Reply-To: <BANLkTimxdvjSU=gDXMLYLb17jDwamWDSbw@mail.gmail.com>
References: <4E0500EE.9040506@499x.com>
	<BANLkTimxdvjSU=gDXMLYLb17jDwamWDSbw@mail.gmail.com>
Message-ID: <4E08DD6A.5040306@499x.com>

I guess part of my confusion here was due to the term "raw data" 
mentioned in conjunction with the .gsm extension on the wiki page 
below... but actually gsm is a compressed format.

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session

So, what is the best "compressed" format to use for recording voice 
(that is available as a direct recording format inside freeswitch)? 
There are tons of formats listed when I do "show file", but I tried a 
few and they are also giving me large files like the wav extension did. 
(au, for example)

Even though the PCM/Wave format is preferred for voice quality, when 
we're talking about a 10:1 compression ratio, if the sound quality is 
still acceptable, I'd rather just record directly into the compressed 
format.  We're talking about ~10- 20 minute recordings that will need to 
be transferred over the internet to a third party.

On 6/24/2011 6:31 PM, Michael Collins wrote:
> I would caution you to consider adding disk space before you try to 
> compress all your recordings. The 16 bit SLIN that FS normally puts in 
> your wave files are pretty easy to handle, whether playing back in a 
> FS session, or encoding for playback on some other device.
>
> An alternative might be to use lame to convert them to MP3's or 
> ogg/vorbis files. If you look on the main FS conf call page you'll see 
> I have the weekly recordings in multiple formats. 
> (http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls)
>
> Here are some stats for last Wednesday's call. Note that I record wave 
> files in 48kHz then use sox to downsample to 16kHz wave, then I 
> convert that 16kHz file into MP3 and Vorbis (in an ogg container). 
> Here's what the results look like:
>
> <2831>:ls -1s conf_call_2011-06-15.*
>  18736 conf_call_2011-06-15.mp3
>  23044 conf_call_2011-06-15.ogg
> 199756 conf_call_2011-06-15.wav
>
> <2832>:file conf_call_2011-06-15.mp3
> conf_call_2011-06-15.mp3: MPEG ADTS, layer III, v2,  24 kBits, 16 kHz, 
> Monaural
>
> <2833>:file conf_call_2011-06-15.ogg
> conf_call_2011-06-15.ogg: Ogg data, Vorbis audio, mono, 16000 Hz, 
> ~48000 bps, created by: Xiph.Org libVorbis I
>
> <2834>:file conf_call_2011-06-15.wav
> conf_call_2011-06-15.wav: RIFF (little-endian) data, WAVE audio, 
> Microsoft PCM, 16 bit, mono 16000 Hz
>
> Note that the file sizes are in 1K blocks.
>
> So, bottom line is this: if you have the disk space then use wave. If 
> you don't have disk space for wave then get some! :D If you REALLY 
> need to use a different format then choose something like MP3 or 
> Vorbis for long-term storage.
>
> -MC
>
> On Fri, Jun 24, 2011 at 2:26 PM, Wes <wes-fs at 499x.com 
> <mailto:wes-fs at 499x.com>> wrote:
>
>     In my tests, if I record a call in .wav format, a 10 second file is
>     about 177,000 bytes, while a 10 second .gsm file is 17,000 bytes.
>
>     I  then used sox to convert the .gsm file to a .wav file, and it
>     stayed
>     at around 17,000 bytes.  So, is the default recording format for .wav
>     using a higher sample rate? vs the default conversion format for
>     the sox
>     tool?
>
>     checking the file type using "file" I see that the larger one is:
>     RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono
>     8000 Hz
>
>     and the wav created by sox via the default conversion from .gsm is:
>     RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz
>
>     So apparently the larger wav file is 16 bit... how are these recording
>     parameters controlled?  Can I set it to record directly into the
>     smaller
>     wav format? Or will I have to run sox on every file...
>
>     _______________________________________________
>     Join us at ClueCon 2011, Aug 9-11, Chicago
>     http://www.cluecon.com 877-7-4ACLUE
>
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>
>
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> http://www.cluecon.com 877-7-4ACLUE
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From bryansmart at bryansmart.com  Tue Jun 28 00:00:01 2011
From: bryansmart at bryansmart.com (Bryan Smart)
Date: Mon, 27 Jun 2011 16:00:01 -0400
Subject: [Freeswitch-users] Proper prompt gain/level
In-Reply-To: <4E088A24.7010002@coppice.org>
References: <FE081E27-8D62-4715-B7AA-64432598E013@bryansmart.com>
	<0D3AEFF37C1643B79CE3F7A7EE9EAEC3@dell9400>
	<66E81300-46E9-4C81-B2E0-123BCADDCAEB@bryansmart.com>
	<4E088A24.7010002@coppice.org>
Message-ID: <3081AB39-6EF0-4C43-9AC1-7D43FCD378CD@bryansmart.com>

I don't want to maximize peaks all the way to 0DB. I just wondered why a low level like -16DB was used. When I'd previously created prompts on Asterisk systems, I used -6DB as a peak (50% of max gain for the channel).

I wasn't aware of any regulations about prompt level. If there is a standardized level, that is what I'd like to use. Perhaps the louder systems are disregarding standards? Does anyone have links to such info? I've been unable to find anything definitive, only opinions.

Listen to TellMe (+1-800-555-8355). It's at least 3X the gain of the default FS prompts. Is TellMe in error? I've called them through both FS and Asterisk, using Vitelity and Callcentric, so I'm fairly sure that I'm not being mislead by a switch or ITSP boosting the gain. I always felt their level was strong and intelligible, without sounding overwhelming.

One point that caught my attention, though, is that you said -16DB for both average and peak power. Average and peak power don't come out the same, though.

So that we can talk about something concrete, consider conference/32000/conf-enter_conf_pin.wav. Its peak power is -15.7DB. However, its average power (RMS) is -31.3DB! -31DB is profoundly quiet. If its average power is boosted to -16DB, then the peak power is now around -2DB. As long as peak power is less than 0DB, then the audio won't clip, but it might be too loud for comfort. I previously used -6DB for a peak, as I couldn't find any real guidelines regarding levels, and -6 sounded good to me.

Maybe FS is lower than it should be. Maybe other services are louder than they should be. If FS should be louder, though, I'd like to help to change the levels up-stream, rather than locally reprocessing the prompts.

So, is this a personal judgement case, or are their standards available that can be consulted?

Bryan

On Jun 27, 2011, at 9:48 AM, Steve Underwood wrote:

> On 06/27/2011 07:11 AM, Bryan Smart wrote:
>> I have tools to batch-process audio files. I just was not sure that regaining all of the prompt files was the best approach. I figured that the gain must have been reduced so dramatically for some sort of reason (to avoid clipping in some situation, to work better with the internal resampling, etc).
>> 
>> What AGC do you mean? I know that AGC has recently been added to conferencing, but the level of the prompts is a system-wide situation. As far as I know, there isn't AGC that can be applied on every channel, and, even if there was, there would surely be a processing hit, so the goal would be to avoid needing it, right?
>> 
>> The root problem, at least for me, is this. I need to add voice prompts and other audio for an IVR. I can't simply normalize all of my prompts to 0DB, as, even though they don't distort, they're so loud when compared to the stock prompts, they'll blow the phone out of my hand. To match them to the stock prompts, I must normalize them to around -16DB. I can do that, but it seems very wrong. At -16DB, nearly 85% of the potential gain of the channel is lost.
> -16dBM0 or -16dBOv, and average or peak burst power? -16dBOv for the 
> average power is about where you want a voice prompt to be. In some 
> juristictions you could be in breach of a regulation or two if you set 
> the level higher than that on the PSTN. Why would you set a voice prompt 
> to 0dB? It will be clipping like crazy.
>> Try this... With the demo IVR (5000), add this before the sleep command in the dialplan.
>> 
>> <action application="set_audio_level" data="write 4"/>
>> 
>> That is the max gain boost available for a channel. The prompts should be really clipping with that much amplification, but they don't clip at all. At -16DB, you could literally amplify them to 6 times their native level without distorting. Native level is too low. Once I realized this, it became clear to me why Freeswitch sounded more quiet than Asterisk, at least when working with recorded prompts.
>> 
>> I suppose I could use set_audio_level on every last call, but I'm sure that real-time amplification, like AGC, is another processor drain that builds up with lots of calls. Besides, it seems weird to dramatically reduce the level of audio, and then waste cycles amplifying it back up in real-time.
>> 
>> Bryan
> Steve
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
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From bryansmart at bryansmart.com  Tue Jun 28 00:03:47 2011
From: bryansmart at bryansmart.com (Bryan Smart)
Date: Mon, 27 Jun 2011 16:03:47 -0400
Subject: [Freeswitch-users] Phrase macros in conferences
In-Reply-To: <9CE9932A-AD27-4183-9C2E-75148116A23B@tech21.com>
References: <FE02E1C7-64EA-4E85-AC33-B491AB2FF555@bryansmart.com>
	<BANLkTikRKyF8eppAtqd6DbQqRH29JLnQ7w@mail.gmail.com>
	<93AE9C49-C6F6-44E6-BC7E-5A31AC659C4C@bryansmart.com>
	<9CE9932A-AD27-4183-9C2E-75148116A23B@tech21.com>
Message-ID: <22CBC366-3CD3-4429-80B6-961AF1566D1C@bryansmart.com>

Yes, but not sure how I'd do that. In the conference profile, I can play a single sound file for an enter/leave. I can't join two files with "!" to have them play in series, like "name.wav!conference/conf-enter.wav".

Bryan

On Jun 27, 2011, at 3:37 PM, Brad Mina wrote:

You could have users prompted to record their name to get around the first issue and play it back before "... Has joined the conference" is played back.

Sent from my iPhone

On Jun 27, 2011, at 12:05 PM, Bryan Smart <bryansmart at bryansmart.com<mailto:bryansmart at bryansmart.com>> wrote:

Thanks, Michael. At least I know that it is a limitation, and not me overlooking something. :) I have 3 or 4 years on Asterisk, but only months on Freeswitch, so I'm always concerned that I haven't dug enough before posting requests.

For phrases inside conferences, I imagine 3 cases.

First are dynamic announcements, such as those that announce when someone joins or leaves a conference. It is helpful if the joining or leaving party can be identified. I imagined combining TTS for the name, along with a pre-recorded fragment like conference/conf-has_joined.wav. The result would be like "John has entered the conference." Right now, I must use entirely TTS for such announcements. I can't use "!" to concatenate files, and I can't defer to a phrase macro to string TTS and recordings together.

Second applies to localization. In a dynamic announcement, it might not be appropriate for the noun to be spoken first in languages other than English. Right now, the conference's prompt language must be customized by creating a different conference profile that specifies the correct sound base for the language. That could mean that the number of profiles in use has to be multiplied by the number of languages that should be supported. I know that the profile can omit the sound base, and the sound base of the first person to join will be used, but, particularly in a situation that concatenates TTS and audio, that can be worse. If the first person to join automatically selects sound files for a language, but the concatenation order is different for that language, then any rules in that profile that concatenate TTS and recordings to render announcements could result in strange outcomes. As far as I've been able to discover, the phrase system is the appropriate Freeswitch way to localize announcements, so it would probably be best for Conference to work that way also, rather than depending on per-case localization being handled with multiple profiles. If phrases were supported, then a single conference profile could play both static and dynamic prompts, and could automatically adjust to any language where phrase macros are present on the system.

Last are cases when multiple types of notification are desired. For example, notifications might use a combination of cues and speech. An audio cue serves a similar function to a visual icon: attracts attention and conveys general meaning. In a situation like a conference, there might be soft audio cues for joining, leaving, when a new person has the floor, etc. When one of those events takes place, the cue plays first to attract attention and set the listener's expectations regarding the type of spoken information that will follow. Next, an announcement is heard with details about the event. "<cue> John has left  the conference."






On Jun 27, 2011, at 11:25 AM, Michael Collins wrote:

Phrase macros are not supported inside mod_conference. If you look inside mod_conference.c you will see that it has its own functions for injecting audio into the conference and it does not support phrase macros. You can open up a feature request on Jira. I honestly don't know how easy/difficult this would be, so you may consider offering a bounty to sweeten the deal.

Curious: what types of phrases are you trying to play into the conference?

-MC

On Sun, Jun 26, 2011 at 4:57 PM, Bryan Smart <<mailto:bryansmart at bryansmart.com>bryansmart at bryansmart.com<mailto:bryansmart at bryansmart.com>> wrote:
I'm attempting to use phrase macros in mod_conference, but I'm not having success. Perhaps they aren't supported?

I have a working phrase macro. In the dialplan, both

<action application="phrase" data="my_phrase"/>

and

<action application="playback" data="phrase:my_phrase"/>

Work just fine.

The wiki says that, in conference.conf.xml, entries like pin-sound accept, for their value, file paths, strings prefixed with "say" for TTS, and strings prefixed with "tone_stream". This looks almost the same as what can be passed to the playback application in the dialplan. When I try to prefix a string with "phrase", it doesn't work.

<param name="pin-sound" value="phrase:my_phrase"/>

The console tells me "unknown file format". The console still tells me this if I make up phrase names

<param name="pin-sound" value="phrase:not_a_real_phrase"/>

So it is complaining about the syntax or capability, rather than an unrecognized format of an actual file.

If mod_conference doesn't support phrase macros, is this a near-future feature? Phrases would seem to be necessary to ocalize announcements and play announcements that are stitched together from multiple files.

Bryan


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From david.ponzone at ipeva.fr  Tue Jun 28 00:26:00 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 27 Jun 2011 22:26:00 +0200
Subject: [Freeswitch-users] Reject SIP registrations
In-Reply-To: <BANLkTikQoYWjH8Nsp20Lg4wsy2O0A3L=4Q@mail.gmail.com>
References: <BANLkTikU6fRYwmUTxzsSHyWs8raukLx4jw@mail.gmail.com>
	<BANLkTimq4YjhjX0vgCt3fsmBcAZRC1OZ-g@mail.gmail.com>
	<BANLkTikQoYWjH8Nsp20Lg4wsy2O0A3L=4Q@mail.gmail.com>
Message-ID: <53E1375B-A8D6-4C1A-A1DD-055A3527673D@ipeva.fr>

The interesting question is then: why are you able to register without password, if this feature is not enabled on the profile...
Perhaps you should recap your config once more, and put the relevant files on PB.

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

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

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




Le 27/06/2011 ? 20:36, Kurtis Heimerl a ?crit :

> That would explain why removing them didn't do anything!
> 
> Thanks.
> 
> On Mon, Jun 27, 2011 at 6:25 AM, Steven Ayre <steveayre at gmail.com> wrote:
>> Just so you know...
>> 
>>      <param name="accept-blind-reg" value="true"/>
>>      <param name="accept-blind-auth" value="true"/>
>> 
>> These will have no effect in the user directory. They only apply to SIP
>> profiles.
>> 
>> -Steve
>> 
>> 
>> 
>> On 27 June 2011 02:23, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
>>> 
>>> Hello FS Users!
>>> 
>>> I'm trying to create the following setup. When a user registers, if
>>> they register on a known account (lets say X), they do not need a
>>> password. X's registration is immediately OK'd, and everything is
>>> great. I've gotten that working using the ACL. The IP address of our
>>> SIP clients are added through cidr and the clients do not need to give
>>> passwords.
>>> 
>>> However, for some reason, if another account that does not exist in
>>> the directory (let's say Y) registers, FS returns with a 200 OK,
>>> instead of rejecting Y. I'm trying to figure out why this is the case,
>>> and how to remedy that fact.
>>> 
>>> I have the following line in my internal.xml file, which I had assumed
>>> would force this function:
>>>   <!-- Force the user and auth-user to match. -->
>>>   <param name="inbound-reg-force-matching-username" value="true"/>
>>> 
>>> However, it does not work. In my directory, each individual account as
>>> the following lines:
>>>  <user id="1303">
>>>    <params>
>>>      <param name="accept-blind-reg" value="true"/>
>>>      <param name="accept-blind-auth" value="true"/>
>>>      <param name="vm-password" value="1000"/>
>>>    </params>
>>> 
>>> Though I've found that removing it (from all users in the directory)
>>> doesn't help.
>>> 
>>> I'm primarily concerned with the line in internal.xml; it seems
>>> possible that the fact that we do not have an auth-user (because we do
>>> not require auth) means that this won't work. However, I have yet to
>>> test that hypothesis. The ACL has been the most confusing aspect of
>>> this installation, with a lot of undocumented aspects, and I get the
>>> nagging feeling this is another. I could very well be wrong though.
>>> 
>>> Thanks for any direction.
>>> 
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
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>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
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>> 
>> 
> 
> _______________________________________________
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From david.ponzone at ipeva.fr  Tue Jun 28 00:29:54 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 27 Jun 2011 22:29:54 +0200
Subject: [Freeswitch-users] .wav vs .gsm file sizes for recording calls.
In-Reply-To: <4E08DD6A.5040306@499x.com>
References: <4E0500EE.9040506@499x.com>
	<BANLkTimxdvjSU=gDXMLYLb17jDwamWDSbw@mail.gmail.com>
	<4E08DD6A.5040306@499x.com>
Message-ID: <D279722B-D927-4F78-8557-3A8CEAAE5516@ipeva.fr>

You should really consider recording in SLIN, and then, batch compressing that later (during off-peak hours) to MP3 if you really need it to be smaller.
Any efficient compression takes CPU, and you may want to avoid doing that live on your server.

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

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

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




Le 27/06/2011 ? 21:43, Wes a ?crit :

> I guess part of my confusion here was due to the term "raw data" mentioned in conjunction with the .gsm extension on the wiki page below... but actually gsm is a compressed format.
> 
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session
> 
> So, what is the best "compressed" format to use for recording voice (that is available as a direct recording format inside freeswitch)? There are tons of formats listed when I do "show file", but I tried a few and they are also giving me large files like the wav extension did. (au, for example)
> 
> Even though the PCM/Wave format is preferred for voice quality, when we're talking about a 10:1 compression ratio, if the sound quality is still acceptable, I'd rather just record directly into the compressed format.  We're talking about ~10- 20 minute recordings that will need to be transferred over the internet to a third party.
> 
> On 6/24/2011 6:31 PM, Michael Collins wrote:
>> 
>> I would caution you to consider adding disk space before you try to compress all your recordings. The 16 bit SLIN that FS normally puts in your wave files are pretty easy to handle, whether playing back in a FS session, or encoding for playback on some other device. 
>> 
>> An alternative might be to use lame to convert them to MP3's or ogg/vorbis files. If you look on the main FS conf call page you'll see I have the weekly recordings in multiple formats. (http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls)
>> 
>> Here are some stats for last Wednesday's call. Note that I record wave files in 48kHz then use sox to downsample to 16kHz wave, then I convert that 16kHz file into MP3 and Vorbis (in an ogg container). Here's what the results look like:
>> 
>> <2831>:ls -1s conf_call_2011-06-15.*
>>  18736 conf_call_2011-06-15.mp3
>>  23044 conf_call_2011-06-15.ogg
>> 199756 conf_call_2011-06-15.wav
>> 
>> <2832>:file conf_call_2011-06-15.mp3 
>> conf_call_2011-06-15.mp3: MPEG ADTS, layer III, v2,  24 kBits, 16 kHz, Monaural
>> 
>> <2833>:file conf_call_2011-06-15.ogg 
>> conf_call_2011-06-15.ogg: Ogg data, Vorbis audio, mono, 16000 Hz, ~48000 bps, created by: Xiph.Org libVorbis I
>> 
>> <2834>:file conf_call_2011-06-15.wav 
>> conf_call_2011-06-15.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 16000 Hz
>> 
>> Note that the file sizes are in 1K blocks.
>> 
>> So, bottom line is this: if you have the disk space then use wave. If you don't have disk space for wave then get some! :D If you REALLY need to use a different format then choose something like MP3 or Vorbis for long-term storage. 
>> 
>> -MC
>> 
>> On Fri, Jun 24, 2011 at 2:26 PM, Wes <wes-fs at 499x.com> wrote:
>> In my tests, if I record a call in .wav format, a 10 second file is
>> about 177,000 bytes, while a 10 second .gsm file is 17,000 bytes.
>> 
>> I  then used sox to convert the .gsm file to a .wav file, and it stayed
>> at around 17,000 bytes.  So, is the default recording format for .wav
>> using a higher sample rate? vs the default conversion format for the sox
>> tool?
>> 
>> checking the file type using "file" I see that the larger one is:
>> RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
>> 
>> and the wav created by sox via the default conversion from .gsm is:
>> RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz
>> 
>> So apparently the larger wav file is 16 bit... how are these recording
>> parameters controlled?  Can I set it to record directly into the smaller
>> wav format? Or will I have to run sox on every file...
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From kheimerl at cs.berkeley.edu  Tue Jun 28 00:30:02 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Mon, 27 Jun 2011 13:30:02 -0700
Subject: [Freeswitch-users] Reject SIP registrations
In-Reply-To: <53E1375B-A8D6-4C1A-A1DD-055A3527673D@ipeva.fr>
References: <BANLkTikU6fRYwmUTxzsSHyWs8raukLx4jw@mail.gmail.com>
	<BANLkTimq4YjhjX0vgCt3fsmBcAZRC1OZ-g@mail.gmail.com>
	<BANLkTikQoYWjH8Nsp20Lg4wsy2O0A3L=4Q@mail.gmail.com>
	<53E1375B-A8D6-4C1A-A1DD-055A3527673D@ipeva.fr>
Message-ID: <BANLkTinrXFE4NYFM4EC7nZqa7jhT-FSLZg@mail.gmail.com>

It's enabled in the acl.conf.xml file, using CIDR.

What conf files do you consider relevant? acl.conf.xml, internal.xml,
a profile or two, anything else?

On Mon, Jun 27, 2011 at 1:26 PM, David Ponzone <david.ponzone at ipeva.fr> wrote:
> The interesting question is then: why are you able to register without
> password, if this feature is not enabled on the profile...
> Perhaps you should recap your config once more, and put the relevant files
> on PB.
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 27/06/2011 ? 20:36, Kurtis Heimerl a ?crit :
>
> That would explain why removing them didn't do anything!
>
> Thanks.
>
> On Mon, Jun 27, 2011 at 6:25 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
> Just so you know...
>
> ? ?? <param name="accept-blind-reg" value="true"/>
>
> ? ? ?<param name="accept-blind-auth" value="true"/>
>
> These will have no effect in the user directory. They only apply to SIP
>
> profiles.
>
> -Steve
>
>
>
> On 27 June 2011 02:23, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
>
> Hello FS Users!
>
> I'm trying to create the following setup. When a user registers, if
>
> they register on a known account (lets say X), they do not need a
>
> password. X's registration is immediately OK'd, and everything is
>
> great. I've gotten that working using the ACL. The IP address of our
>
> SIP clients are added through cidr and the clients do not need to give
>
> passwords.
>
> However, for some reason, if another account that does not exist in
>
> the directory (let's say Y) registers, FS returns with a 200 OK,
>
> instead of rejecting Y. I'm trying to figure out why this is the case,
>
> and how to remedy that fact.
>
> I have the following line in my internal.xml file, which I had assumed
>
> would force this function:
>
> ? <!-- Force the user and auth-user to match. -->
>
> ? <param name="inbound-reg-force-matching-username" value="true"/>
>
> However, it does not work. In my directory, each individual account as
>
> the following lines:
>
> ?<user id="1303">
>
> ? ?<params>
>
> ? ? ?<param name="accept-blind-reg" value="true"/>
>
> ? ? ?<param name="accept-blind-auth" value="true"/>
>
> ? ? ?<param name="vm-password" value="1000"/>
>
> ? ?</params>
>
> Though I've found that removing it (from all users in the directory)
>
> doesn't help.
>
> I'm primarily concerned with the line in internal.xml; it seems
>
> possible that the fact that we do not have an auth-user (because we do
>
> not require auth) means that this won't work. However, I have yet to
>
> test that hypothesis. The ACL has been the most confusing aspect of
>
> this installation, with a lot of undocumented aspects, and I get the
>
> nagging feeling this is another. I could very well be wrong though.
>
> Thanks for any direction.
>
> _______________________________________________
>
> Join us at ClueCon 2011, Aug 9-11, Chicago
>
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
> _______________________________________________
>
> Join us at ClueCon 2011, Aug 9-11, Chicago
>
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From wes-fs at 499x.com  Tue Jun 28 00:45:44 2011
From: wes-fs at 499x.com (Wes)
Date: Mon, 27 Jun 2011 15:45:44 -0500
Subject: [Freeswitch-users] .wav vs .gsm file sizes for recording calls.
In-Reply-To: <D279722B-D927-4F78-8557-3A8CEAAE5516@ipeva.fr>
References: <4E0500EE.9040506@499x.com>	<BANLkTimxdvjSU=gDXMLYLb17jDwamWDSbw@mail.gmail.com>	<4E08DD6A.5040306@499x.com>
	<D279722B-D927-4F78-8557-3A8CEAAE5516@ipeva.fr>
Message-ID: <4E08EBF8.9040207@499x.com>

I'm sorry, but what is the SLIN format?  I tried that extension and it's 
invalid.  I searched the wiki and found nothing.

On 6/27/2011 3:29 PM, David Ponzone wrote:
> You should really consider recording in SLIN, and then, batch 
> compressing that later (during off-peak hours) to MP3 if you really 
> need it to be smaller.
> Any efficient compression takes CPU, and you may want to avoid doing 
> that live on your server.
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service ClientIPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
> diffusion non autoris?e est interdite. Tout message ?lectronique est 
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au 
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
> n'?tes pas destinataire de ce message, merci de le d?truire 
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 27/06/2011 ? 21:43, Wes a ?crit :
>
>> I guess part of my confusion here was due to the term "raw data" 
>> mentioned in conjunction with the .gsm extension on the wiki page 
>> below... but actually gsm is a compressed format.
>>
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session
>>
>> So, what is the best "compressed" format to use for recording voice 
>> (that is available as a direct recording format inside freeswitch)? 
>> There are tons of formats listed when I do "show file", but I tried a 
>> few and they are also giving me large files like the wav extension 
>> did. (au, for example)
>>
>> Even though the PCM/Wave format is preferred for voice quality, when 
>> we're talking about a 10:1 compression ratio, if the sound quality is 
>> still acceptable, I'd rather just record directly into the compressed 
>> format.  We're talking about ~10- 20 minute recordings that will need 
>> to be transferred over the internet to a third party.
>>
>> On 6/24/2011 6:31 PM, Michael Collins wrote:
>>> I would caution you to consider adding disk space before you try to 
>>> compress all your recordings. The 16 bit SLIN that FS normally puts 
>>> in your wave files are pretty easy to handle, whether playing back 
>>> in a FS session, or encoding for playback on some other device.
>>>
>>> An alternative might be to use lame to convert them to MP3's or 
>>> ogg/vorbis files. If you look on the main FS conf call page you'll 
>>> see I have the weekly recordings in multiple formats. 
>>> (http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls)
>>>
>>> Here are some stats for last Wednesday's call. Note that I record 
>>> wave files in 48kHz then use sox to downsample to 16kHz wave, then I 
>>> convert that 16kHz file into MP3 and Vorbis (in an ogg container). 
>>> Here's what the results look like:
>>>
>>> <2831>:ls -1s conf_call_2011-06-15.*
>>>  18736 conf_call_2011-06-15.mp3
>>>  23044 conf_call_2011-06-15.ogg
>>> 199756 conf_call_2011-06-15.wav
>>>
>>> <2832>:file conf_call_2011-06-15.mp3
>>> conf_call_2011-06-15.mp3: MPEG ADTS, layer III, v2,  24 kBits, 16 
>>> kHz, Monaural
>>>
>>> <2833>:file conf_call_2011-06-15.ogg
>>> conf_call_2011-06-15.ogg: Ogg data, Vorbis audio, mono, 16000 Hz, 
>>> ~48000 bps, created by: Xiph.Org <http://Xiph.Org> libVorbis I
>>>
>>> <2834>:file conf_call_2011-06-15.wav
>>> conf_call_2011-06-15.wav: RIFF (little-endian) data, WAVE audio, 
>>> Microsoft PCM, 16 bit, mono 16000 Hz
>>>
>>> Note that the file sizes are in 1K blocks.
>>>
>>> So, bottom line is this: if you have the disk space then use wave. 
>>> If you don't have disk space for wave then get some! :D If you 
>>> REALLY need to use a different format then choose something like MP3 
>>> or Vorbis for long-term storage.
>>>
>>> -MC
>>>
>>> On Fri, Jun 24, 2011 at 2:26 PM, Wes <wes-fs at 499x.com 
>>> <mailto:wes-fs at 499x.com>> wrote:
>>>
>>>     In my tests, if I record a call in .wav format, a 10 second file is
>>>     about 177,000 bytes, while a 10 second .gsm file is 17,000 bytes.
>>>
>>>     I  then used sox to convert the .gsm file to a .wav file, and it
>>>     stayed
>>>     at around 17,000 bytes.  So, is the default recording format for
>>>     .wav
>>>     using a higher sample rate? vs the default conversion format for
>>>     the sox
>>>     tool?
>>>
>>>     checking the file type using "file" I see that the larger one is:
>>>     RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit,
>>>     mono 8000 Hz
>>>
>>>     and the wav created by sox via the default conversion from .gsm is:
>>>     RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz
>>>
>>>     So apparently the larger wav file is 16 bit... how are these
>>>     recording
>>>     parameters controlled?  Can I set it to record directly into the
>>>     smaller
>>>     wav format? Or will I have to run sox on every file...
>>>
>>>     _______________________________________________
>>>     Join us at ClueCon 2011, Aug 9-11, Chicago
>>>     http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>
>>>     FreeSWITCH-users mailing list
>>>     FreeSWITCH-users at lists.freeswitch.org
>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>>
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com  877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org 
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From jmoran at secureachsystems.com  Tue Jun 28 01:52:47 2011
From: jmoran at secureachsystems.com (Jason Moran)
Date: Mon, 27 Jun 2011 17:52:47 -0400
Subject: [Freeswitch-users] UniMRCP Server as daemon won't connect RTSP
Message-ID: <361E98F99D3CC3439EED59BC1924ED6950815F@SERVER2003.SecuReachSystems.local>

I've been playing around with IVONA's new unimrcp plugin with unimrcp
1.0.0 (r1725) on OpenSuse 11.4

 

Anyway, when I start it up with the following, it stays in the
foreground of my SSH w/ the following command:

./unimrcpserver

That works! However, since it's in the foreground when I close the SSH
window, it kills the process.

 

If I use the so-called daemon mode (-d or --daemon) it says it is going
into daemon mode, but FS will immediately return that it "Failed to
Connect to RTSP Server..." at the IP:port I specified. The unimrcpserver
process is running, but doesn't seem to respond to anything.

./unimrcpserver -d

./unimrcpserver --daemon

 

If I attempt to background it by either using "&" or ctrl-z it says [1]+
Stopped  ./unimrcpserver and FS will successfully make the RTSP
connection but then nothing will happen. Mod_unimrcp will spit out a
warning about "MRCP session has not opened after 5000 ms"

./unimrcpserver &

Those are hard to kill. But when I kill -9 it then FS will finally
remove the MRCP handle and tell me that it couldn't allocate the speech
engine.

 

Lastly, I can nohup it, but then unimrcpserver eats up 95%+ of the CPU
(instead of 1-3% as it does when I have it in the foreground), which it
does not do when it runs in any other modes. It also makes a very, very
large nohup.out file that keeps on growing. Even if I tell it to not
log.

nohup ./unimrcpserver &

nohup ./unimrcpserver -o 0 &

 

Ideas??
Thanks,

Jason Moran

 

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From anthony.minessale at gmail.com  Tue Jun 28 02:07:08 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 27 Jun 2011 17:07:08 -0500
Subject: [Freeswitch-users] Proper prompt gain/level
In-Reply-To: <3081AB39-6EF0-4C43-9AC1-7D43FCD378CD@bryansmart.com>
References: <FE081E27-8D62-4715-B7AA-64432598E013@bryansmart.com>
	<0D3AEFF37C1643B79CE3F7A7EE9EAEC3@dell9400>
	<66E81300-46E9-4C81-B2E0-123BCADDCAEB@bryansmart.com>
	<4E088A24.7010002@coppice.org>
	<3081AB39-6EF0-4C43-9AC1-7D43FCD378CD@bryansmart.com>
Message-ID: <BANLkTikTK3JW=jDFNqkZqPxhic_bzL=uCQ@mail.gmail.com>

Have a look at the 48khz versions of the files, those should be the originals.
Someone runs a batch sox command on them to get them to the other levels.

I will say I find the volume of many phone systems overdriven and I
think the goal was to keep the sound files in the neighborhood of what
a live person sounds like with no clipping.  I am quite sure nobody
was looking at the specific dB levels when generating them.

I am not really overly picky about what level we distribute at and I
would defer it to the majority of users to see if there is a
perception of the volume being too low in our sound files.

If there is an overwhelming agreement that the volume is too low, I
would be willing to adjust it if you are interested in taking
ownership of the task.



On Mon, Jun 27, 2011 at 3:00 PM, Bryan Smart <bryansmart at bryansmart.com> wrote:
> I don't want to maximize peaks all the way to 0DB. I just wondered why a low level like -16DB was used. When I'd previously created prompts on Asterisk systems, I used -6DB as a peak (50% of max gain for the channel).
>
> I wasn't aware of any regulations about prompt level. If there is a standardized level, that is what I'd like to use. Perhaps the louder systems are disregarding standards? Does anyone have links to such info? I've been unable to find anything definitive, only opinions.
>
> Listen to TellMe (+1-800-555-8355). It's at least 3X the gain of the default FS prompts. Is TellMe in error? I've called them through both FS and Asterisk, using Vitelity and Callcentric, so I'm fairly sure that I'm not being mislead by a switch or ITSP boosting the gain. I always felt their level was strong and intelligible, without sounding overwhelming.
>
> One point that caught my attention, though, is that you said -16DB for both average and peak power. Average and peak power don't come out the same, though.
>
> So that we can talk about something concrete, consider conference/32000/conf-enter_conf_pin.wav. Its peak power is -15.7DB. However, its average power (RMS) is -31.3DB! -31DB is profoundly quiet. If its average power is boosted to -16DB, then the peak power is now around -2DB. As long as peak power is less than 0DB, then the audio won't clip, but it might be too loud for comfort. I previously used -6DB for a peak, as I couldn't find any real guidelines regarding levels, and -6 sounded good to me.
>
> Maybe FS is lower than it should be. Maybe other services are louder than they should be. If FS should be louder, though, I'd like to help to change the levels up-stream, rather than locally reprocessing the prompts.
>
> So, is this a personal judgement case, or are their standards available that can be consulted?
>
> Bryan
>
> On Jun 27, 2011, at 9:48 AM, Steve Underwood wrote:
>
>> On 06/27/2011 07:11 AM, Bryan Smart wrote:
>>> I have tools to batch-process audio files. I just was not sure that regaining all of the prompt files was the best approach. I figured that the gain must have been reduced so dramatically for some sort of reason (to avoid clipping in some situation, to work better with the internal resampling, etc).
>>>
>>> What AGC do you mean? I know that AGC has recently been added to conferencing, but the level of the prompts is a system-wide situation. As far as I know, there isn't AGC that can be applied on every channel, and, even if there was, there would surely be a processing hit, so the goal would be to avoid needing it, right?
>>>
>>> The root problem, at least for me, is this. I need to add voice prompts and other audio for an IVR. I can't simply normalize all of my prompts to 0DB, as, even though they don't distort, they're so loud when compared to the stock prompts, they'll blow the phone out of my hand. To match them to the stock prompts, I must normalize them to around -16DB. I can do that, but it seems very wrong. At -16DB, nearly 85% of the potential gain of the channel is lost.
>> -16dBM0 or -16dBOv, and average or peak burst power? -16dBOv for the
>> average power is about where you want a voice prompt to be. In some
>> juristictions you could be in breach of a regulation or two if you set
>> the level higher than that on the PSTN. Why would you set a voice prompt
>> to 0dB? It will be clipping like crazy.
>>> Try this... With the demo IVR (5000), add this before the sleep command in the dialplan.
>>>
>>> <action application="set_audio_level" data="write 4"/>
>>>
>>> That is the max gain boost available for a channel. The prompts should be really clipping with that much amplification, but they don't clip at all. At -16DB, you could literally amplify them to 6 times their native level without distorting. Native level is too low. Once I realized this, it became clear to me why Freeswitch sounded more quiet than Asterisk, at least when working with recorded prompts.
>>>
>>> I suppose I could use set_audio_level on every last call, but I'm sure that real-time amplification, like AGC, is another processor drain that builds up with lots of calls. Besides, it seems weird to dramatically reduce the level of audio, and then waste cycles amplifying it back up in real-time.
>>>
>>> Bryan
>> Steve
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From steveayre at gmail.com  Tue Jun 28 02:10:32 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 27 Jun 2011 23:10:32 +0100
Subject: [Freeswitch-users] .wav vs .gsm file sizes for recording calls.
In-Reply-To: <4E08DD6A.5040306@499x.com>
References: <4E0500EE.9040506@499x.com>
	<BANLkTimxdvjSU=gDXMLYLb17jDwamWDSbw@mail.gmail.com>
	<4E08DD6A.5040306@499x.com>
Message-ID: <A9469BBE-0F9B-4889-B3BD-BCC96487E985@gmail.com>

Its not just voice quality. It's also CPU.

If you're using a compressed format, you're doing transcoding step. FS needs to decompress the file to L16 so it can compress it to the codec. Storing as wav containing l16 would avoid that. You can also use mod_native_file to compress the audio file to each codec format so that you don't need to transcode at all at the time because it's already done, although that does mean a file for each codec so more space needed.

Steve on iPhone

On 27 Jun 2011, at 20:43, Wes <wes-fs at 499x.com> wrote:

> I guess part of my confusion here was due to the term "raw data" mentioned in conjunction with the .gsm extension on the wiki page below... but actually gsm is a compressed format.
> 
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session
> 
> So, what is the best "compressed" format to use for recording voice (that is available as a direct recording format inside freeswitch)? There are tons of formats listed when I do "show file", but I tried a few and they are also giving me large files like the wav extension did. (au, for example)
> 
> Even though the PCM/Wave format is preferred for voice quality, when we're talking about a 10:1 compression ratio, if the sound quality is still acceptable, I'd rather just record directly into the compressed format.  We're talking about ~10- 20 minute recordings that will need to be transferred over the internet to a third party.
> 
> On 6/24/2011 6:31 PM, Michael Collins wrote:
>> 
>> I would caution you to consider adding disk space before you try to compress all your recordings. The 16 bit SLIN that FS normally puts in your wave files are pretty easy to handle, whether playing back in a FS session, or encoding for playback on some other device. 
>> 
>> An alternative might be to use lame to convert them to MP3's or ogg/vorbis files. If you look on the main FS conf call page you'll see I have the weekly recordings in multiple formats. (http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls)
>> 
>> Here are some stats for last Wednesday's call. Note that I record       wave files in 48kHz then use sox to downsample to 16kHz wave, then I convert that 16kHz file into MP3 and Vorbis (in an ogg container). Here's what the results look like:
>> 
>> <2831>:ls -1s conf_call_2011-06-15.*
>>  18736 conf_call_2011-06-15.mp3
>>  23044 conf_call_2011-06-15.ogg
>> 199756 conf_call_2011-06-15.wav
>> 
>> <2832>:file conf_call_2011-06-15.mp3 
>> conf_call_2011-06-15.mp3: MPEG ADTS, layer III, v2,  24 kBits, 16 kHz, Monaural
>> 
>> <2833>:file conf_call_2011-06-15.ogg 
>> conf_call_2011-06-15.ogg: Ogg data, Vorbis audio, mono, 16000 Hz, ~48000 bps, created by: Xiph.Org libVorbis I
>> 
>> <2834>:file conf_call_2011-06-15.wav 
>> conf_call_2011-06-15.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 16000 Hz
>> 
>> Note that the file sizes are in 1K blocks.
>> 
>> So, bottom line is this: if you have the disk space then use wave. If you don't have disk space for wave then get some! :D If you REALLY need to use a different format then choose something like MP3 or Vorbis for long-term storage. 
>> 
>> -MC
>> 
>> On Fri, Jun 24, 2011 at 2:26 PM, Wes <wes-fs at 499x.com> wrote:
>> In my tests, if I record a call in .wav format, a 10 second file is
>> about 177,000 bytes, while a 10 second .gsm file is 17,000 bytes.
>> 
>> I  then used sox to convert the .gsm file to a .wav file, and it stayed
>> at around 17,000 bytes.  So, is the default recording format for .wav
>> using a higher sample rate? vs the default conversion format for the sox
>> tool?
>> 
>> checking the file type using "file" I see that the larger one is:
>> RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
>> 
>> and the wav created by sox via the default conversion from .gsm is:
>> RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz
>> 
>> So apparently the larger wav file is 16 bit... how are these recording
>> parameters controlled?  Can I set it to record directly into the smaller
>> wav format? Or will I have to run sox on every file...
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From bill.evergreen at gmail.com  Mon Jun 27 20:15:53 2011
From: bill.evergreen at gmail.com (bill evergreen)
Date: Mon, 27 Jun 2011 18:15:53 +0200
Subject: [Freeswitch-users] does FreeSWITCH care about beeing connected to
	an IPv4- or IPv6-ISP ?
Message-ID: <BANLkTikb0DQE8cKxBn_kPuzH47QjBmCXQg@mail.gmail.com>

Hello,

I intend to setup a little FreeSWITCH test-environment.  Does FreeSWITCH
care about beeing connected to an IPv4- OR IPv6-isp (it should be possible
to operate the same environment under IPv4 OR IPv6) ?


Thank's a lot for any feedback!

Bill



PS
I am new to FreeSWITCH :-)
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From freeswitch at ml102.pinguin.uni.cc  Tue Jun 28 00:05:04 2011
From: freeswitch at ml102.pinguin.uni.cc (Al Bogner)
Date: Mon, 27 Jun 2011 22:05:04 +0200
Subject: [Freeswitch-users] SIP/2.0 403 Forbidden
Message-ID: <20110627220504.377cc403@ml102.pinguin.uni.cc>

I am doing my first steps with freeswitch. 

I have installed on an Ubuntu 11.04 server:

ii  freeswitch                           1.0.7~20110603-0natty6  
ii  freeswitch-codec-passthru-g7231      1.0.7~20110603-0natty6  
ii  freeswitch-codec-passthru-g729       1.0.7~20110603-0natty6  
ii  freeswitch-lang-de                   1.0.7~20110603-0natty6  
ii  freeswitch-lang-en                   1.0.7~20110603-0natty6


Following the tutorial at
http://www.onlinesolution.co.nz/viewtopic.php?t=102
I modified /opt/freeswitch/conf/directory/default/1000.xml

<include>
  <user id="1000">
    <params>
      <param name="password" value="siptest"/>
      <param name="vm-password" value="siptest"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,local"/>
      <variable name="accountcode" value="1000"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="Extension 1000"/>
      <variable name="effective_caller_id_number" value="1000"/>
      <variable name="outbound_caller_id_name"
value="$${outbound_caller_name}"/> <variable
name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
<variable name="callgroup" value="techsupport"/> </variables>
  </user>
</include>


Then I tried to connect to the server from another machine:


+++ 27-6-2011 17:07:30.279796 INFO SIP ::send_sip_udp
Send to: udp:192.168.2.100:5060
REGISTER sip:192.168.2.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.108;rport;branch=z9hG4bKcvkohckk
Max-Forwards: 70
To: "FS Admin" <sip:1000 at 192.168.2.100>
From: "FS Admin" <sip:1000 at 192.168.2.100>;tag=nyyvz
Call-ID: zmjxppampyrrgxk at client.local.tld
CSeq: 645 REGISTER
Contact: <sip:1000 at 192.168.2.108>;expires=3600
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
User-Agent: Twinkle/1.4.2 Content-Length: 0


---

+++ 27-6-2011 17:07:30.281799 INFO SIP ::process_sip_msg
Received from: udp:192.168.2.100:5060
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP
192.168.2.108;received=192.168.2.108;rport=5060;branch=z9hG4bKcvkohckk
To: "FS Admin" <sip:1000 at 192.168.2.100>;tag=iairy From: "FS Admin"
<sip:1000 at 192.168.2.100>;tag=nyyvz Call-ID:
zmjxppampyrrgxk at client.local.tld CSeq: 645 REGISTER
Server: Twinkle/1.4.2
Content-Length: 0


What could be wrong, so I can't connect?

Al


From dave at dchorton.com  Tue Jun 28 01:12:48 2011
From: dave at dchorton.com (Dave Horton)
Date: Mon, 27 Jun 2011 17:12:48 -0400
Subject: [Freeswitch-users] building rpm doesn't get me sounds directory
Message-ID: <E6F838A7-68C0-4893-B176-E79BB150500F@dchorton.com>

I'm building an rpm using the freeswitch.spec in the top-level directory, and it creates a usable rpm for me except that when I install it on a target machine, I don't get the sounds sub-directory and therefore voicemail doesn't work.  Do I need to make some changes to the spec file or the rpm build script in order to get an rpm that will install all of the sounds/recordings? 

From kheimerl at cs.berkeley.edu  Tue Jun 28 03:11:33 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Mon, 27 Jun 2011 16:11:33 -0700
Subject: [Freeswitch-users] Reject SIP registrations
In-Reply-To: <BANLkTinrXFE4NYFM4EC7nZqa7jhT-FSLZg@mail.gmail.com>
References: <BANLkTikU6fRYwmUTxzsSHyWs8raukLx4jw@mail.gmail.com>
	<BANLkTimq4YjhjX0vgCt3fsmBcAZRC1OZ-g@mail.gmail.com>
	<BANLkTikQoYWjH8Nsp20Lg4wsy2O0A3L=4Q@mail.gmail.com>
	<53E1375B-A8D6-4C1A-A1DD-055A3527673D@ipeva.fr>
	<BANLkTinrXFE4NYFM4EC7nZqa7jhT-FSLZg@mail.gmail.com>
Message-ID: <BANLkTinoOwUsjNPSrdXUB4pMkq3LXU1v7w@mail.gmail.com>

Anyhow, here are those three config files:

internal.xml : http://pastebin.freeswitch.org/16609
acl.conf.xml : http://pastebin.freeswitch.org/16610
1300.xml : http://pastebin.freeswitch.org/16611

If anything else could help, I'd love to share it.

The basic story, so far as I see, is that I allow specific IPs through
the ACL. Somehow this is allowing ANY SIP username to register, rather
than just those defined (such as 1300). Any help would be appreciated.

On Mon, Jun 27, 2011 at 1:30 PM, Kurtis Heimerl
<kheimerl at cs.berkeley.edu> wrote:
> It's enabled in the acl.conf.xml file, using CIDR.
>
> What conf files do you consider relevant? acl.conf.xml, internal.xml,
> a profile or two, anything else?
>
> On Mon, Jun 27, 2011 at 1:26 PM, David Ponzone <david.ponzone at ipeva.fr> wrote:
>> The interesting question is then: why are you able to register without
>> password, if this feature is not enabled on the profile...
>> Perhaps you should recap your config once more, and put the relevant files
>> on PB.
>> David Ponzone ?Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel: ? ? ?01 74 03 18 97
>> gsm: ? 06 66 98 76 34
>> Service Client?IPeva
>> tel: ? ? ?0811 46 26 26
>> www.ipeva.fr? -? ?www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>> Le 27/06/2011 ? 20:36, Kurtis Heimerl a ?crit :
>>
>> That would explain why removing them didn't do anything!
>>
>> Thanks.
>>
>> On Mon, Jun 27, 2011 at 6:25 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> Just so you know...
>>
>> ? ?? <param name="accept-blind-reg" value="true"/>
>>
>> ? ? ?<param name="accept-blind-auth" value="true"/>
>>
>> These will have no effect in the user directory. They only apply to SIP
>>
>> profiles.
>>
>> -Steve
>>
>>
>>
>> On 27 June 2011 02:23, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
>>
>> Hello FS Users!
>>
>> I'm trying to create the following setup. When a user registers, if
>>
>> they register on a known account (lets say X), they do not need a
>>
>> password. X's registration is immediately OK'd, and everything is
>>
>> great. I've gotten that working using the ACL. The IP address of our
>>
>> SIP clients are added through cidr and the clients do not need to give
>>
>> passwords.
>>
>> However, for some reason, if another account that does not exist in
>>
>> the directory (let's say Y) registers, FS returns with a 200 OK,
>>
>> instead of rejecting Y. I'm trying to figure out why this is the case,
>>
>> and how to remedy that fact.
>>
>> I have the following line in my internal.xml file, which I had assumed
>>
>> would force this function:
>>
>> ? <!-- Force the user and auth-user to match. -->
>>
>> ? <param name="inbound-reg-force-matching-username" value="true"/>
>>
>> However, it does not work. In my directory, each individual account as
>>
>> the following lines:
>>
>> ?<user id="1303">
>>
>> ? ?<params>
>>
>> ? ? ?<param name="accept-blind-reg" value="true"/>
>>
>> ? ? ?<param name="accept-blind-auth" value="true"/>
>>
>> ? ? ?<param name="vm-password" value="1000"/>
>>
>> ? ?</params>
>>
>> Though I've found that removing it (from all users in the directory)
>>
>> doesn't help.
>>
>> I'm primarily concerned with the line in internal.xml; it seems
>>
>> possible that the fact that we do not have an auth-user (because we do
>>
>> not require auth) means that this won't work. However, I have yet to
>>
>> test that hypothesis. The ACL has been the most confusing aspect of
>>
>> this installation, with a lot of undocumented aspects, and I get the
>>
>> nagging feeling this is another. I could very well be wrong though.
>>
>> Thanks for any direction.
>>
>> _______________________________________________
>>
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>>
>> FreeSWITCH-users at lists.freeswitch.org
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>>
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>>
>> FreeSWITCH-users at lists.freeswitch.org
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>


From kheimerl at cs.berkeley.edu  Tue Jun 28 03:12:59 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Mon, 27 Jun 2011 16:12:59 -0700
Subject: [Freeswitch-users] Reject SIP registrations
In-Reply-To: <BANLkTinoOwUsjNPSrdXUB4pMkq3LXU1v7w@mail.gmail.com>
References: <BANLkTikU6fRYwmUTxzsSHyWs8raukLx4jw@mail.gmail.com>
	<BANLkTimq4YjhjX0vgCt3fsmBcAZRC1OZ-g@mail.gmail.com>
	<BANLkTikQoYWjH8Nsp20Lg4wsy2O0A3L=4Q@mail.gmail.com>
	<53E1375B-A8D6-4C1A-A1DD-055A3527673D@ipeva.fr>
	<BANLkTinrXFE4NYFM4EC7nZqa7jhT-FSLZg@mail.gmail.com>
	<BANLkTinoOwUsjNPSrdXUB4pMkq3LXU1v7w@mail.gmail.com>
Message-ID: <BANLkTikaVD+3Xi_z4SxGrz06GzktmrfHbw@mail.gmail.com>

One of those links got screwed up...

Anyhow, here are those three config files:

internal.xml : http://pastebin.freeswitch.org/16609

acl.conf.xml : http://pastebin.freeswitch.org/16610

1300.xml : http://pastebin.freeswitch.org/16611

If anything else could help, I'd love to share it.

The basic story, so far as I see, is that I allow specific IPs through
the ACL. Somehow this is allowing ANY SIP username to register, rather
than just those defined (such as 1300). Any help would be appreciated.

On Mon, Jun 27, 2011 at 4:11 PM, Kurtis Heimerl
<kheimerl at cs.berkeley.edu> wrote:
> Anyhow, here are those three config files:
>
> internal.xml : http://pastebin.freeswitch.org/16609
> acl.conf.xml : http://pastebin.freeswitch.org/16610
> 1300.xml : http://pastebin.freeswitch.org/16611
>
> If anything else could help, I'd love to share it.
>
> The basic story, so far as I see, is that I allow specific IPs through
> the ACL. Somehow this is allowing ANY SIP username to register, rather
> than just those defined (such as 1300). Any help would be appreciated.
>
> On Mon, Jun 27, 2011 at 1:30 PM, Kurtis Heimerl
> <kheimerl at cs.berkeley.edu> wrote:
>> It's enabled in the acl.conf.xml file, using CIDR.
>>
>> What conf files do you consider relevant? acl.conf.xml, internal.xml,
>> a profile or two, anything else?
>>
>> On Mon, Jun 27, 2011 at 1:26 PM, David Ponzone <david.ponzone at ipeva.fr> wrote:
>>> The interesting question is then: why are you able to register without
>>> password, if this feature is not enabled on the profile...
>>> Perhaps you should recap your config once more, and put the relevant files
>>> on PB.
>>> David Ponzone ?Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel: ? ? ?01 74 03 18 97
>>> gsm: ? 06 66 98 76 34
>>> Service Client?IPeva
>>> tel: ? ? ?0811 46 26 26
>>> www.ipeva.fr? -? ?www.ipeva-studio.com
>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>
>>>
>>>
>>> Le 27/06/2011 ? 20:36, Kurtis Heimerl a ?crit :
>>>
>>> That would explain why removing them didn't do anything!
>>>
>>> Thanks.
>>>
>>> On Mon, Jun 27, 2011 at 6:25 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>>
>>> Just so you know...
>>>
>>> ? ?? <param name="accept-blind-reg" value="true"/>
>>>
>>> ? ? ?<param name="accept-blind-auth" value="true"/>
>>>
>>> These will have no effect in the user directory. They only apply to SIP
>>>
>>> profiles.
>>>
>>> -Steve
>>>
>>>
>>>
>>> On 27 June 2011 02:23, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
>>>
>>> Hello FS Users!
>>>
>>> I'm trying to create the following setup. When a user registers, if
>>>
>>> they register on a known account (lets say X), they do not need a
>>>
>>> password. X's registration is immediately OK'd, and everything is
>>>
>>> great. I've gotten that working using the ACL. The IP address of our
>>>
>>> SIP clients are added through cidr and the clients do not need to give
>>>
>>> passwords.
>>>
>>> However, for some reason, if another account that does not exist in
>>>
>>> the directory (let's say Y) registers, FS returns with a 200 OK,
>>>
>>> instead of rejecting Y. I'm trying to figure out why this is the case,
>>>
>>> and how to remedy that fact.
>>>
>>> I have the following line in my internal.xml file, which I had assumed
>>>
>>> would force this function:
>>>
>>> ? <!-- Force the user and auth-user to match. -->
>>>
>>> ? <param name="inbound-reg-force-matching-username" value="true"/>
>>>
>>> However, it does not work. In my directory, each individual account as
>>>
>>> the following lines:
>>>
>>> ?<user id="1303">
>>>
>>> ? ?<params>
>>>
>>> ? ? ?<param name="accept-blind-reg" value="true"/>
>>>
>>> ? ? ?<param name="accept-blind-auth" value="true"/>
>>>
>>> ? ? ?<param name="vm-password" value="1000"/>
>>>
>>> ? ?</params>
>>>
>>> Though I've found that removing it (from all users in the directory)
>>>
>>> doesn't help.
>>>
>>> I'm primarily concerned with the line in internal.xml; it seems
>>>
>>> possible that the fact that we do not have an auth-user (because we do
>>>
>>> not require auth) means that this won't work. However, I have yet to
>>>
>>> test that hypothesis. The ACL has been the most confusing aspect of
>>>
>>> this installation, with a lot of undocumented aspects, and I get the
>>>
>>> nagging feeling this is another. I could very well be wrong though.
>>>
>>> Thanks for any direction.
>>>
>>> _______________________________________________
>>>
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>>
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>


From david.ponzone at ipeva.fr  Tue Jun 28 03:28:32 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 28 Jun 2011 01:28:32 +0200
Subject: [Freeswitch-users] .wav vs .gsm file sizes for recording calls.
In-Reply-To: <A9469BBE-0F9B-4889-B3BD-BCC96487E985@gmail.com>
References: <4E0500EE.9040506@499x.com>
	<BANLkTimxdvjSU=gDXMLYLb17jDwamWDSbw@mail.gmail.com>
	<4E08DD6A.5040306@499x.com>
	<A9469BBE-0F9B-4889-B3BD-BCC96487E985@gmail.com>
Message-ID: <3C36C8D8-79C7-46F8-BB09-E9321E8B5425@ipeva.fr>

Steven,

AFAIR, one of the recording app (not sure which) does not support mod_native_file, meaning for instance you can't dump your call using codec XXX to a raw file.
That's a quite old limitation, so perhaps it's not an issue anymore.

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

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

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




Le 28/06/2011 ? 00:10, Steven Ayre a ?crit :

> Its not just voice quality. It's also CPU.
> 
> If you're using a compressed format, you're doing transcoding step. FS needs to decompress the file to L16 so it can compress it to the codec. Storing as wav containing l16 would avoid that. You can also use mod_native_file to compress the audio file to each codec format so that you don't need to transcode at all at the time because it's already done, although that does mean a file for each codec so more space needed.
> 
> Steve on iPhone
> 
> On 27 Jun 2011, at 20:43, Wes <wes-fs at 499x.com> wrote:
> 
>> I guess part of my confusion here was due to the term "raw data" mentioned in conjunction with the .gsm extension on the wiki page below... but actually gsm is a compressed format.
>> 
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session
>> 
>> So, what is the best "compressed" format to use for recording voice (that is available as a direct recording format inside freeswitch)? There are tons of formats listed when I do "show file", but I tried a few and they are also giving me large files like the wav extension did. (au, for example)
>> 
>> Even though the PCM/Wave format is preferred for voice quality, when we're talking about a 10:1 compression ratio, if the sound quality is still acceptable, I'd rather just record directly into the compressed format.  We're talking about ~10- 20 minute recordings that will need to be transferred over the internet to a third party.
>> 
>> On 6/24/2011 6:31 PM, Michael Collins wrote:
>>> 
>>> I would caution you to consider adding disk space before you try to compress all your recordings. The 16 bit SLIN that FS normally puts in your wave files are pretty easy to handle, whether playing back in a FS session, or encoding for playback on some other device. 
>>> 
>>> An alternative might be to use lame to convert them to MP3's or ogg/vorbis files. If you look on the main FS conf call page you'll see I have the weekly recordings in multiple formats. (http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls)
>>> 
>>> Here are some stats for last Wednesday's call. Note that I record wave files in 48kHz then use sox to downsample to 16kHz wave, then I convert that 16kHz file into MP3 and Vorbis (in an ogg container). Here's what the results look like:
>>> 
>>> <2831>:ls -1s conf_call_2011-06-15.*
>>>  18736 conf_call_2011-06-15.mp3
>>>  23044 conf_call_2011-06-15.ogg
>>> 199756 conf_call_2011-06-15.wav
>>> 
>>> <2832>:file conf_call_2011-06-15.mp3 
>>> conf_call_2011-06-15.mp3: MPEG ADTS, layer III, v2,  24 kBits, 16 kHz, Monaural
>>> 
>>> <2833>:file conf_call_2011-06-15.ogg 
>>> conf_call_2011-06-15.ogg: Ogg data, Vorbis audio, mono, 16000 Hz, ~48000 bps, created by: Xiph.Org libVorbis I
>>> 
>>> <2834>:file conf_call_2011-06-15.wav 
>>> conf_call_2011-06-15.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 16000 Hz
>>> 
>>> Note that the file sizes are in 1K blocks.
>>> 
>>> So, bottom line is this: if you have the disk space then use wave. If you don't have disk space for wave then get some! :D If you REALLY need to use a different format then choose something like MP3 or Vorbis for long-term storage. 
>>> 
>>> -MC
>>> 
>>> On Fri, Jun 24, 2011 at 2:26 PM, Wes <wes-fs at 499x.com> wrote:
>>> In my tests, if I record a call in .wav format, a 10 second file is
>>> about 177,000 bytes, while a 10 second .gsm file is 17,000 bytes.
>>> 
>>> I  then used sox to convert the .gsm file to a .wav file, and it stayed
>>> at around 17,000 bytes.  So, is the default recording format for .wav
>>> using a higher sample rate? vs the default conversion format for the sox
>>> tool?
>>> 
>>> checking the file type using "file" I see that the larger one is:
>>> RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
>>> 
>>> and the wav created by sox via the default conversion from .gsm is:
>>> RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz
>>> 
>>> So apparently the larger wav file is 16 bit... how are these recording
>>> parameters controlled?  Can I set it to record directly into the smaller
>>> wav format? Or will I have to run sox on every file...
>>> 
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From msc at freeswitch.org  Tue Jun 28 03:32:35 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 27 Jun 2011 16:32:35 -0700
Subject: [Freeswitch-users] Proper prompt gain/level
In-Reply-To: <BANLkTikTK3JW=jDFNqkZqPxhic_bzL=uCQ@mail.gmail.com>
References: <FE081E27-8D62-4715-B7AA-64432598E013@bryansmart.com>
	<0D3AEFF37C1643B79CE3F7A7EE9EAEC3@dell9400>
	<66E81300-46E9-4C81-B2E0-123BCADDCAEB@bryansmart.com>
	<4E088A24.7010002@coppice.org>
	<3081AB39-6EF0-4C43-9AC1-7D43FCD378CD@bryansmart.com>
	<BANLkTikTK3JW=jDFNqkZqPxhic_bzL=uCQ@mail.gmail.com>
Message-ID: <BANLkTimSuFpBXwANu8qc2q24z-xNYdcJnw@mail.gmail.com>

On Mon, Jun 27, 2011 at 3:07 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Have a look at the 48khz versions of the files, those should be the
> originals.
> Someone runs a batch sox command on them to get them to the other levels.
>

Just for the record, the sox command used to normalize the sounds is:
sox -v 0.2 $file_in -r $rate -c 1 $file_out

The -v format is the "volume" format and if I read the sox man page
correctly it means a "linear amplitude adjustment". What I can't tell from
the man page is whether .2 means "reduce by 20%" or "reduce to 20%" or
something else. In any case, like Tony says, we are not picky about this as
long as it's not pointlessly loud.

-MC
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From msc at freeswitch.org  Tue Jun 28 03:42:16 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 27 Jun 2011 16:42:16 -0700
Subject: [Freeswitch-users] .wav vs .gsm file sizes for recording calls.
In-Reply-To: <4E08EBF8.9040207@499x.com>
References: <4E0500EE.9040506@499x.com>
	<BANLkTimxdvjSU=gDXMLYLb17jDwamWDSbw@mail.gmail.com>
	<4E08DD6A.5040306@499x.com>
	<D279722B-D927-4F78-8557-3A8CEAAE5516@ipeva.fr>
	<4E08EBF8.9040207@499x.com>
Message-ID: <BANLkTin5nFqjn6f8CCg=p23UEaVXTzq4AA@mail.gmail.com>

On Mon, Jun 27, 2011 at 1:45 PM, Wes <wes-fs at 499x.com> wrote:

> **
> I'm sorry, but what is the SLIN format?  I tried that extension and it's
> invalid.  I searched the wiki and found nothing.
>
http://en.wikipedia.org/wiki/Linear_pulse-code_modulation

There's lots of names for it. Just think of SLIN as "the format normally
stored in a .wav file." (That's an imperfect definition but it will help you
get over the mental hump.)

In summary, just record to .wav files and x-code later when there are more
system resources available. You could also use "nice" or something like that
to prevent the x-coding process from grabbing all your system resources.

-MC


>
> On 6/27/2011 3:29 PM, David Ponzone wrote:
>
> You should really consider recording in SLIN, and then, batch compressing
> that later (during off-peak hours) to MP3 if you really need it to be
> smaller.
> Any efficient compression takes CPU, and you may want to avoid doing that
> live on your server.
>
>   David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
>  Service Client IPeva
>  tel:      0811 46 26 26
>  www.ipeva.fr  -   www.ipeva-studio.com
>
>  *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
>  Le 27/06/2011 ? 21:43, Wes a ?crit :
>
>  I guess part of my confusion here was due to the term "raw data"
> mentioned in conjunction with the .gsm extension on the wiki page below...
> but actually gsm is a compressed format.
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session
>
> So, what is the best "compressed" format to use for recording voice (that
> is available as a direct recording format inside freeswitch)? There are tons
> of formats listed when I do "show file", but I tried a few and they are also
> giving me large files like the wav extension did. (au, for example)
>
> Even though the PCM/Wave format is preferred for voice quality, when we're
> talking about a 10:1 compression ratio, if the sound quality is still
> acceptable, I'd rather just record directly into the compressed format.
> We're talking about ~10- 20 minute recordings that will need to be
> transferred over the internet to a third party.
>
> On 6/24/2011 6:31 PM, Michael Collins wrote:
>
> I would caution you to consider adding disk space before you try to
> compress all your recordings. The 16 bit SLIN that FS normally puts in your
> wave files are pretty easy to handle, whether playing back in a FS session,
> or encoding for playback on some other device.
>
> An alternative might be to use lame to convert them to MP3's or ogg/vorbis
> files. If you look on the main FS conf call page you'll see I have the
> weekly recordings in multiple formats. (
> http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls)
>
> Here are some stats for last Wednesday's call. Note that I record wave
> files in 48kHz then use sox to downsample to 16kHz wave, then I convert that
> 16kHz file into MP3 and Vorbis (in an ogg container). Here's what the
> results look like:
>
> <2831>:ls -1s conf_call_2011-06-15.*
>  18736 conf_call_2011-06-15.mp3
>  23044 conf_call_2011-06-15.ogg
> 199756 conf_call_2011-06-15.wav
>
> <2832>:file conf_call_2011-06-15.mp3
> conf_call_2011-06-15.mp3: MPEG ADTS, layer III, v2,  24 kBits, 16 kHz,
> Monaural
>
> <2833>:file conf_call_2011-06-15.ogg
> conf_call_2011-06-15.ogg: Ogg data, Vorbis audio, mono, 16000 Hz, ~48000
> bps, created by: Xiph.Org libVorbis I
>
> <2834>:file conf_call_2011-06-15.wav
> conf_call_2011-06-15.wav: RIFF (little-endian) data, WAVE audio, Microsoft
> PCM, 16 bit, mono 16000 Hz
>
> Note that the file sizes are in 1K blocks.
>
> So, bottom line is this: if you have the disk space then use wave. If you
> don't have disk space for wave then get some! :D If you REALLY need to use a
> different format then choose something like MP3 or Vorbis for long-term
> storage.
>
> -MC
>
> On Fri, Jun 24, 2011 at 2:26 PM, Wes <wes-fs at 499x.com> wrote:
>
>> In my tests, if I record a call in .wav format, a 10 second file is
>> about 177,000 bytes, while a 10 second .gsm file is 17,000 bytes.
>>
>> I  then used sox to convert the .gsm file to a .wav file, and it stayed
>> at around 17,000 bytes.  So, is the default recording format for .wav
>> using a higher sample rate? vs the default conversion format for the sox
>> tool?
>>
>> checking the file type using "file" I see that the larger one is:
>> RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
>>
>> and the wav created by sox via the default conversion from .gsm is:
>> RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz
>>
>> So apparently the larger wav file is 16 bit... how are these recording
>> parameters controlled?  Can I set it to record directly into the smaller
>> wav format? Or will I have to run sox on every file...
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>  _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Tue Jun 28 03:55:08 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 28 Jun 2011 01:55:08 +0200
Subject: [Freeswitch-users] SIP/2.0 403 Forbidden
In-Reply-To: <20110627220504.377cc403@ml102.pinguin.uni.cc>
References: <20110627220504.377cc403@ml102.pinguin.uni.cc>
Message-ID: <77CE8968-A79D-4310-964D-E4952F1D436C@ipeva.fr>

Al,

please, show us the content of the following files:
conf/directory/default.xml
conf/vars.xml

May you also tell us if you see a message in fs_cli when you try to register ?

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

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

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




Le 27/06/2011 ? 22:05, Al Bogner a ?crit :

> I am doing my first steps with freeswitch. 
> 
> I have installed on an Ubuntu 11.04 server:
> 
> ii  freeswitch                           1.0.7~20110603-0natty6  
> ii  freeswitch-codec-passthru-g7231      1.0.7~20110603-0natty6  
> ii  freeswitch-codec-passthru-g729       1.0.7~20110603-0natty6  
> ii  freeswitch-lang-de                   1.0.7~20110603-0natty6  
> ii  freeswitch-lang-en                   1.0.7~20110603-0natty6
> 
> 
> Following the tutorial at
> http://www.onlinesolution.co.nz/viewtopic.php?t=102
> I modified /opt/freeswitch/conf/directory/default/1000.xml
> 
> <include>
>  <user id="1000">
>    <params>
>      <param name="password" value="siptest"/>
>      <param name="vm-password" value="siptest"/>
>    </params>
>    <variables>
>      <variable name="toll_allow" value="domestic,local"/>
>      <variable name="accountcode" value="1000"/>
>      <variable name="user_context" value="default"/>
>      <variable name="effective_caller_id_name" value="Extension 1000"/>
>      <variable name="effective_caller_id_number" value="1000"/>
>      <variable name="outbound_caller_id_name"
> value="$${outbound_caller_name}"/> <variable
> name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
> <variable name="callgroup" value="techsupport"/> </variables>
>  </user>
> </include>
> 
> 
> Then I tried to connect to the server from another machine:
> 
> 
> +++ 27-6-2011 17:07:30.279796 INFO SIP ::send_sip_udp
> Send to: udp:192.168.2.100:5060
> REGISTER sip:192.168.2.100 SIP/2.0
> Via: SIP/2.0/UDP 192.168.2.108;rport;branch=z9hG4bKcvkohckk
> Max-Forwards: 70
> To: "FS Admin" <sip:1000 at 192.168.2.100>
> From: "FS Admin" <sip:1000 at 192.168.2.100>;tag=nyyvz
> Call-ID: zmjxppampyrrgxk at client.local.tld
> CSeq: 645 REGISTER
> Contact: <sip:1000 at 192.168.2.108>;expires=3600
> Allow:
> INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
> User-Agent: Twinkle/1.4.2 Content-Length: 0
> 
> 
> ---
> 
> +++ 27-6-2011 17:07:30.281799 INFO SIP ::process_sip_msg
> Received from: udp:192.168.2.100:5060
> SIP/2.0 403 Forbidden
> Via: SIP/2.0/UDP
> 192.168.2.108;received=192.168.2.108;rport=5060;branch=z9hG4bKcvkohckk
> To: "FS Admin" <sip:1000 at 192.168.2.100>;tag=iairy From: "FS Admin"
> <sip:1000 at 192.168.2.100>;tag=nyyvz Call-ID:
> zmjxppampyrrgxk at client.local.tld CSeq: 645 REGISTER
> Server: Twinkle/1.4.2
> Content-Length: 0
> 
> 
> What could be wrong, so I can't connect?
> 
> Al
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From steveu at coppice.org  Tue Jun 28 05:31:00 2011
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 28 Jun 2011 09:31:00 +0800
Subject: [Freeswitch-users] Proper prompt gain/level
In-Reply-To: <3081AB39-6EF0-4C43-9AC1-7D43FCD378CD@bryansmart.com>
References: <FE081E27-8D62-4715-B7AA-64432598E013@bryansmart.com>	<0D3AEFF37C1643B79CE3F7A7EE9EAEC3@dell9400>	<66E81300-46E9-4C81-B2E0-123BCADDCAEB@bryansmart.com>	<4E088A24.7010002@coppice.org>
	<3081AB39-6EF0-4C43-9AC1-7D43FCD378CD@bryansmart.com>
Message-ID: <4E092ED4.60605@coppice.org>

On 06/28/2011 04:00 AM, Bryan Smart wrote:
> I don't want to maximize peaks all the way to 0DB. I just wondered why a low level like -16DB was used. When I'd previously created prompts on Asterisk systems, I used -6DB as a peak (50% of max gain for the channel).
The peaks of voice are general something like 12 to 13dB above the short 
term RMS value. If the -16dB value you quoted is -16dBOv, its 13dB below 
clipping. Your peaks should be close to clipping at -16dB.
> I wasn't aware of any regulations about prompt level. If there is a standardized level, that is what I'd like to use. Perhaps the louder systems are disregarding standards? Does anyone have links to such info? I've been unable to find anything definitive, only opinions.
In many places the regulation says the power on a PSTN line should not 
exceed -13dBm0. That's why -13dBm0 is the target power level for most 
PSTN modems.
> Listen to TellMe (+1-800-555-8355). It's at least 3X the gain of the default FS prompts. Is TellMe in error? I've called them through both FS and Asterisk, using Vitelity and Callcentric, so I'm fairly sure that I'm not being mislead by a switch or ITSP boosting the gain. I always felt their level was strong and intelligible, without sounding overwhelming.
Not everyone bothers to obey regulations these days, and many people do 
love to blast sound into an overloaded highly distorted mess, because 
volume is king. Its not good for clarity, though. What the loudest 
people do is no measure of good engineering. Sadly, this behaviour might 
make people set their levels around the excessively loud signals, so a 
properly adjusted signal sounds too quiet. In the early days of my FAX 
modem work I received many recordings from people who could not get 
reliable results, where the audio was perpetually in clipping, and any 
speech through the channel would have sounded awful. They would 
generally insist that voice was "perfect" on their system. There is a 
serious lack of engineering in most VoIP work.
> One point that caught my attention, though, is that you said -16DB for both average and peak power. Average and peak power don't come out the same, though.
>
> So that we can talk about something concrete, consider conference/32000/conf-enter_conf_pin.wav. Its peak power is -15.7DB. However, its average power (RMS) is -31.3DB! -31DB is profoundly quiet. If its average power is boosted to -16DB, then the peak power is now around -2DB. As long as peak power is less than 0DB, then the audio won't clip, but it might be too loud for comfort. I previously used -6DB for a peak, as I couldn't find any real guidelines regarding levels, and -6 sounded good to me.
You still haven't said whether you are talking dBm0 or dBOv. It makes a 
6dB difference. Also, what do you mean by peak power? If the peaks of 
the short term RMS power are hitting 0dB, the peaks of the waveform will 
be far into clipping. If you are talking about dBOv, then -6dB is only 
3dB from the onset of clipping, and voice will clip a lot. If you are 
talking dBm0, -6dB is 9dB from clipping, and the voice will only clip a 
bit, and maybe not sound too bad. However, clipped voice tends to pass 
through low bit rate codecs worse than clean voice, so you might want to 
keep the clipping down to a really occasional event. Voice codecs have 
at least 12 bits of dynamic range. They are designed to allow a voice to 
bubble along at -30dB with good quality, and burst up to a much higher 
level in the loud bits.
> Maybe FS is lower than it should be. Maybe other services are louder than they should be. If FS should be louder, though, I'd like to help to change the levels up-stream, rather than locally reprocessing the prompts.
>
> So, is this a personal judgement case, or are their standards available that can be consulted?
>
> Bryan
>
> On Jun 27, 2011, at 9:48 AM, Steve Underwood wrote:
>
>> On 06/27/2011 07:11 AM, Bryan Smart wrote:
>>> I have tools to batch-process audio files. I just was not sure that regaining all of the prompt files was the best approach. I figured that the gain must have been reduced so dramatically for some sort of reason (to avoid clipping in some situation, to work better with the internal resampling, etc).
>>>
>>> What AGC do you mean? I know that AGC has recently been added to conferencing, but the level of the prompts is a system-wide situation. As far as I know, there isn't AGC that can be applied on every channel, and, even if there was, there would surely be a processing hit, so the goal would be to avoid needing it, right?
>>>
>>> The root problem, at least for me, is this. I need to add voice prompts and other audio for an IVR. I can't simply normalize all of my prompts to 0DB, as, even though they don't distort, they're so loud when compared to the stock prompts, they'll blow the phone out of my hand. To match them to the stock prompts, I must normalize them to around -16DB. I can do that, but it seems very wrong. At -16DB, nearly 85% of the potential gain of the channel is lost.
>> -16dBM0 or -16dBOv, and average or peak burst power? -16dBOv for the
>> average power is about where you want a voice prompt to be. In some
>> juristictions you could be in breach of a regulation or two if you set
>> the level higher than that on the PSTN. Why would you set a voice prompt
>> to 0dB? It will be clipping like crazy.
>>> Try this... With the demo IVR (5000), add this before the sleep command in the dialplan.
>>>
>>> <action application="set_audio_level" data="write 4"/>
>>>
>>> That is the max gain boost available for a channel. The prompts should be really clipping with that much amplification, but they don't clip at all. At -16DB, you could literally amplify them to 6 times their native level without distorting. Native level is too low. Once I realized this, it became clear to me why Freeswitch sounded more quiet than Asterisk, at least when working with recorded prompts.
>>>
>>> I suppose I could use set_audio_level on every last call, but I'm sure that real-time amplification, like AGC, is another processor drain that builds up with lots of calls. Besides, it seems weird to dramatically reduce the level of audio, and then waste cycles amplifying it back up in real-time.
>>>
>>> Bryan
>> Steve
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From admin at blindi.net  Tue Jun 28 06:20:39 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Tue, 28 Jun 2011 04:20:39 +0200 (CEST)
Subject: [Freeswitch-users] Question why repeat the FS the
 group_confirm_file announcement 3 times?
In-Reply-To: <4E092ED4.60605@coppice.org>
References: <FE081E27-8D62-4715-B7AA-64432598E013@bryansmart.com>
	<0D3AEFF37C1643B79CE3F7A7EE9EAEC3@dell9400>
	<66E81300-46E9-4C81-B2E0-123BCADDCAEB@bryansmart.com>
	<4E088A24.7010002@coppice.org>
	<3081AB39-6EF0-4C43-9AC1-7D43FCD378CD@bryansmart.com>
	<4E092ED4.60605@coppice.org>
Message-ID: <Pine.LNX.4.64.1106280407470.10836@tomback.antagus.de>


Hi all,
i created the following dial plan:
<extension name="call_in_play">
<condition field="destination_number" expression="^498925007676">
  <action application="sched_hangup" data="+3599"/>
  <action application="set" data="group_confirm_file=playback 
$${base_dir}/sound/calls/callparty.wav"/>
   <action application="set" data="group_confirm_key=exec"/>
<action application="transfer" data="0 XML operator1"/>
</condition>
</extension>

Fs play the soundfile 3 times befor the connection
Can i change these please?
thanks.

group_confirm_file

---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From curriegrad2004 at gmail.com  Tue Jun 28 06:25:07 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 27 Jun 2011 19:25:07 -0700
Subject: [Freeswitch-users] does FreeSWITCH care about beeing connected
 to an IPv4- or IPv6-ISP ?
In-Reply-To: <BANLkTikb0DQE8cKxBn_kPuzH47QjBmCXQg@mail.gmail.com>
References: <BANLkTikb0DQE8cKxBn_kPuzH47QjBmCXQg@mail.gmail.com>
Message-ID: <BANLkTi=w8R5qy-AHTTfXO=PYb+JxjieFzw@mail.gmail.com>

FreeSwitch supports both IPv4 and IPv6 protocols via it's Sofia SIP
stack. There is a default internal profile created for the IPv6 side
of things, but then again, you might want to customize that first.

On Mon, Jun 27, 2011 at 9:15 AM, bill evergreen
<bill.evergreen at gmail.com> wrote:
> Hello,
>
> I intend to setup a little FreeSWITCH test-environment.? Does FreeSWITCH
> care about beeing connected to an IPv4- OR IPv6-isp (it should be possible
> to operate the same environment under IPv4 OR IPv6) ?
>
>
> Thank's a lot for any feedback!
>
> Bill
>
>
>
> PS
> I am new to FreeSWITCH :-)
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From admin at blindi.net  Tue Jun 28 06:38:01 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Tue, 28 Jun 2011 04:38:01 +0200 (CEST)
Subject: [Freeswitch-users] How can i drop the outcallcontext from the
 originate callerid?
In-Reply-To: <C8355FEB-5D0C-4C7B-867A-764FB49D1CAD@ipeva.fr>
References: <D4A666272A214C4DBBABFE569CA34CA401DFB8D374@doctorbob.loop.local>
	<BANLkTikbHBgsdZq_yh=RkXbb4O70S4UtTA@mail.gmail.com>
	<4DD547F5.4080504@gmail.com>
	<BANLkTimxdKSkemEYM3x-1+BSES5GKiboJA@mail.gmail.com>
	<BANLkTimxS6GeHU4=3m_JPHTnjr7RBvoqXg@mail.gmail.com>
	<Pine.LNX.4.64.1105230943190.3447@debian>
	<C8355FEB-5D0C-4C7B-867A-764FB49D1CAD@ipeva.fr>
Message-ID: <Pine.LNX.4.64.1106280432170.10836@tomback.antagus.de>

Hi David and michael, sorry for the delay.
I was much busy.
I have fixed my problem.

I insert this action:
<action application="set"
data="rufnummer=${regex(${caller_id_number}|^(.*?)/|%1"/>
    now it works fine for me

thanks for your nice help!
---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From bryansmart at bryansmart.com  Tue Jun 28 07:04:07 2011
From: bryansmart at bryansmart.com (Bryan Smart)
Date: Mon, 27 Jun 2011 23:04:07 -0400
Subject: [Freeswitch-users] Proper prompt gain/level
In-Reply-To: <4E092ED4.60605@coppice.org>
References: <FE081E27-8D62-4715-B7AA-64432598E013@bryansmart.com>
	<0D3AEFF37C1643B79CE3F7A7EE9EAEC3@dell9400>
	<66E81300-46E9-4C81-B2E0-123BCADDCAEB@bryansmart.com>
	<4E088A24.7010002@coppice.org>
	<3081AB39-6EF0-4C43-9AC1-7D43FCD378CD@bryansmart.com>
	<4E092ED4.60605@coppice.org>
Message-ID: <2D33125A-05E0-40DE-92A2-7038940977AB@bryansmart.com>

I think that dBm0 only applies if we are measuring the power on an analog circuit, or at the d/a point of a digital circuit.

I was performing analysis of a digitally encoded audio file, therefore the measurement is dBFS, and 0DBFS is the point where clipping takes place. Anything below 0DBFS does not clip, even though it might seem loud to someone.

I stated that the peak power of the stock prompts are typically -15 to -16 (DBFS), and the short term RMS power is about -32DBFS.

I don't know how to properly evaluate how DBFS will convert to DBm0. I gather that the codec and the d/a converter attenuate the level to some extent, but I don't know how much. If I have a digital file with a 1Khz test tone at -10DBFS, and play that over the pstn, what do I get out in DBm0?

Speaking just in terms of DBFS, -16DB is only about 15% of the potential power available before the audio reaches 0DB, and clips. Perhaps when -16DBFS is put out over the pstn, it is much closer to clipping than it would be in an entirely digital domain. I don't know enough to make that determination. Do you know?

I'm not for clipping and distortion. However, too little power can cause another problem: limited dynamic range and increased dithering artifacts. If audio is quiet on a pstn phone, then the person with the phone might be able to increase the level by turning up the phone's volume, if it has one. However, that raises the noise floor. Companding might make 8-bit channels sound a bit like 14-bit or 15-bit channels in terms of a low noise floor and decreased dithering artifacts, but it is still just an 8-bit channel. Companding hides most of the dithering artifacts in strong signals, but it magnifies them in weak signals. That's why G711 is fairly clear, but will sound scratchy if you put faint signals in to it and try to amplify them back up to normal levels. Over G711 or any pstn call, as you decrease the level of the audio going across, the scratchy dithering static obscures an increasing amount of the audio.

This gets worse if you compound it by stacking codecs. Consider if someone calls your IVR from a cell phone, and you play quiet prompts to them. The audio is first passed through G711, where the low gain means that scratchy dithering artifacts are added. Then, it is encoded to GSM for the cell phone,. GSM uses linear predictive coding, and, being tuned for voice, it is not optimized for having smooth waveforms interrupted periodically with random excursions. Thats part of why cell phone calls sound so bad if there is lots of background noise.

Anyway, clipping is to be avoided, but simply reducing levels dramatically creates other quality problems on a channel that uses companding. You trade distorted audio for scratchy audio. Same thing happened with cassette tapes that used Dolby noise reduction.

Maybe you can help clear up my understanding of how DBFS in a digital file will work out in DBm0 on a pstn line. As things stand, though, I think we have lots of room to increase the level of the prompts before we reach a point where clipping is an issue.

Bryan

On Jun 27, 2011, at 9:31 PM, Steve Underwood wrote:

> On 06/28/2011 04:00 AM, Bryan Smart wrote:
>> I don't want to maximize peaks all the way to 0DB. I just wondered why a low level like -16DB was used. When I'd previously created prompts on Asterisk systems, I used -6DB as a peak (50% of max gain for the channel).
> The peaks of voice are general something like 12 to 13dB above the short 
> term RMS value. If the -16dB value you quoted is -16dBOv, its 13dB below 
> clipping. Your peaks should be close to clipping at -16dB.
>> I wasn't aware of any regulations about prompt level. If there is a standardized level, that is what I'd like to use. Perhaps the louder systems are disregarding standards? Does anyone have links to such info? I've been unable to find anything definitive, only opinions.
> In many places the regulation says the power on a PSTN line should not 
> exceed -13dBm0. That's why -13dBm0 is the target power level for most 
> PSTN modems.
>> Listen to TellMe (+1-800-555-8355). It's at least 3X the gain of the default FS prompts. Is TellMe in error? I've called them through both FS and Asterisk, using Vitelity and Callcentric, so I'm fairly sure that I'm not being mislead by a switch or ITSP boosting the gain. I always felt their level was strong and intelligible, without sounding overwhelming.
> Not everyone bothers to obey regulations these days, and many people do 
> love to blast sound into an overloaded highly distorted mess, because 
> volume is king. Its not good for clarity, though. What the loudest 
> people do is no measure of good engineering. Sadly, this behaviour might 
> make people set their levels around the excessively loud signals, so a 
> properly adjusted signal sounds too quiet. In the early days of my FAX 
> modem work I received many recordings from people who could not get 
> reliable results, where the audio was perpetually in clipping, and any 
> speech through the channel would have sounded awful. They would 
> generally insist that voice was "perfect" on their system. There is a 
> serious lack of engineering in most VoIP work.
>> One point that caught my attention, though, is that you said -16DB for both average and peak power. Average and peak power don't come out the same, though.
>> 
>> So that we can talk about something concrete, consider conference/32000/conf-enter_conf_pin.wav. Its peak power is -15.7DB. However, its average power (RMS) is -31.3DB! -31DB is profoundly quiet. If its average power is boosted to -16DB, then the peak power is now around -2DB. As long as peak power is less than 0DB, then the audio won't clip, but it might be too loud for comfort. I previously used -6DB for a peak, as I couldn't find any real guidelines regarding levels, and -6 sounded good to me.
> You still haven't said whether you are talking dBm0 or dBOv. It makes a 
> 6dB difference. Also, what do you mean by peak power? If the peaks of 
> the short term RMS power are hitting 0dB, the peaks of the waveform will 
> be far into clipping. If you are talking about dBOv, then -6dB is only 
> 3dB from the onset of clipping, and voice will clip a lot. If you are 
> talking dBm0, -6dB is 9dB from clipping, and the voice will only clip a 
> bit, and maybe not sound too bad. However, clipped voice tends to pass 
> through low bit rate codecs worse than clean voice, so you might want to 
> keep the clipping down to a really occasional event. Voice codecs have 
> at least 12 bits of dynamic range. They are designed to allow a voice to 
> bubble along at -30dB with good quality, and burst up to a much higher 
> level in the loud bits.
>> Maybe FS is lower than it should be. Maybe other services are louder than they should be. If FS should be louder, though, I'd like to help to change the levels up-stream, rather than locally reprocessing the prompts.
>> 
>> So, is this a personal judgement case, or are their standards available that can be consulted?
>> 
>> Bryan
>> 
>> On Jun 27, 2011, at 9:48 AM, Steve Underwood wrote:
>> 
>>> On 06/27/2011 07:11 AM, Bryan Smart wrote:
>>>> I have tools to batch-process audio files. I just was not sure that regaining all of the prompt files was the best approach. I figured that the gain must have been reduced so dramatically for some sort of reason (to avoid clipping in some situation, to work better with the internal resampling, etc).
>>>> 
>>>> What AGC do you mean? I know that AGC has recently been added to conferencing, but the level of the prompts is a system-wide situation. As far as I know, there isn't AGC that can be applied on every channel, and, even if there was, there would surely be a processing hit, so the goal would be to avoid needing it, right?
>>>> 
>>>> The root problem, at least for me, is this. I need to add voice prompts and other audio for an IVR. I can't simply normalize all of my prompts to 0DB, as, even though they don't distort, they're so loud when compared to the stock prompts, they'll blow the phone out of my hand. To match them to the stock prompts, I must normalize them to around -16DB. I can do that, but it seems very wrong. At -16DB, nearly 85% of the potential gain of the channel is lost.
>>> -16dBM0 or -16dBOv, and average or peak burst power? -16dBOv for the
>>> average power is about where you want a voice prompt to be. In some
>>> juristictions you could be in breach of a regulation or two if you set
>>> the level higher than that on the PSTN. Why would you set a voice prompt
>>> to 0dB? It will be clipping like crazy.
>>>> Try this... With the demo IVR (5000), add this before the sleep command in the dialplan.
>>>> 
>>>> <action application="set_audio_level" data="write 4"/>
>>>> 
>>>> That is the max gain boost available for a channel. The prompts should be really clipping with that much amplification, but they don't clip at all. At -16DB, you could literally amplify them to 6 times their native level without distorting. Native level is too low. Once I realized this, it became clear to me why Freeswitch sounded more quiet than Asterisk, at least when working with recorded prompts.
>>>> 
>>>> I suppose I could use set_audio_level on every last call, but I'm sure that real-time amplification, like AGC, is another processor drain that builds up with lots of calls. Besides, it seems weird to dramatically reduce the level of audio, and then waste cycles amplifying it back up in real-time.
>>>> 
>>>> Bryan
>>> Steve
>>> 
>>> 
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From bryansmart at bryansmart.com  Tue Jun 28 07:25:54 2011
From: bryansmart at bryansmart.com (Bryan Smart)
Date: Mon, 27 Jun 2011 23:25:54 -0400
Subject: [Freeswitch-users] Proper prompt gain/level
In-Reply-To: <BANLkTimSuFpBXwANu8qc2q24z-xNYdcJnw@mail.gmail.com>
References: <FE081E27-8D62-4715-B7AA-64432598E013@bryansmart.com>
	<0D3AEFF37C1643B79CE3F7A7EE9EAEC3@dell9400>
	<66E81300-46E9-4C81-B2E0-123BCADDCAEB@bryansmart.com>
	<4E088A24.7010002@coppice.org>
	<3081AB39-6EF0-4C43-9AC1-7D43FCD378CD@bryansmart.com>
	<BANLkTikTK3JW=jDFNqkZqPxhic_bzL=uCQ@mail.gmail.com>
	<BANLkTimSuFpBXwANu8qc2q24z-xNYdcJnw@mail.gmail.com>
Message-ID: <1BB0CF88-DA61-4AB6-9CC6-328CCB9846AD@bryansmart.com>

-v is a multiplier. So, -v 0.2 reduces gain to 20% of original.

It is typical to supply professionally recorded audio with peaks boosted to 0DBFS. That is probably how the prompts were originally supplied. It makes since that -v 0.2 would reduce the max power to 20%, around -15 to -16 DBFS.

The FS build process downloads pre-processed versions of these files. Is there somewhere online that I can find the source recordings+script that produces the pre-processed files that are downloaded by the build process?

Bryan

On Jun 27, 2011, at 7:32 PM, Michael Collins wrote:



On Mon, Jun 27, 2011 at 3:07 PM, Anthony Minessale <anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>> wrote:
Have a look at the 48khz versions of the files, those should be the originals.
Someone runs a batch sox command on them to get them to the other levels.

Just for the record, the sox command used to normalize the sounds is:
sox -v 0.2 $file_in -r $rate -c 1 $file_out

The -v format is the "volume" format and if I read the sox man page correctly it means a "linear amplitude adjustment". What I can't tell from the man page is whether .2 means "reduce by 20%" or "reduce to 20%" or something else. In any case, like Tony says, we are not picky about this as long as it's not pointlessly loud.

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
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From michal.bielicki at seventhsignal.de  Tue Jun 28 09:47:57 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Tue, 28 Jun 2011 07:47:57 +0200
Subject: [Freeswitch-users] building rpm doesn't get me sounds directory
In-Reply-To: <E6F838A7-68C0-4893-B176-E79BB150500F@dchorton.com>
References: <E6F838A7-68C0-4893-B176-E79BB150500F@dchorton.com>
Message-ID: <FFEDEAE8-F0E9-4F18-B523-F9307AA318F3@seventhsignal.de>

THere are two spec files, one for freeswitch itself and one for sound files. Did not do a moh spec file yet but the soundfile spec file of course will make a sounds directory too since it installs all sound files in it.

Am 27.06.2011 um 23:12 schrieb Dave Horton:

> I'm building an rpm using the freeswitch.spec in the top-level directory, and it creates a usable rpm for me except that when I install it on a target machine, I don't get the sounds sub-directory and therefore voicemail doesn't work.  Do I need to make some changes to the spec file or the rpm build script in order to get an rpm that will install all of the sounds/recordings? 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Gesch?ftsf?hrer / CEO

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

Amtsgericht Charlottenburg HRA 44413 B
Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de



From peter.olsson at visionutveckling.se  Tue Jun 28 09:57:00 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 28 Jun 2011 07:57:00 +0200
Subject: [Freeswitch-users] Reject SIP registrations
Message-ID: <7E6172B9-85D8-44E7-AA6E-481A8157BF58@visionutveckling.se>

AFAIK this is the intended behaviour.

If allowed through ACL using cidr, the user is treated as authenticated, so no registration is required.

/Peter

----- Reply message -----
Fr?n: "Kurtis Heimerl" <kheimerl at cs.berkeley.edu>
Datum: tis, jun 28, 2011 01:19
Rubrik: [Freeswitch-users] Reject SIP registrations
Till: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

One of those links got screwed up...

Anyhow, here are those three config files:

internal.xml : http://pastebin.freeswitch.org/16609

acl.conf.xml : http://pastebin.freeswitch.org/16610

1300.xml : http://pastebin.freeswitch.org/16611

If anything else could help, I'd love to share it.

The basic story, so far as I see, is that I allow specific IPs through
the ACL. Somehow this is allowing ANY SIP username to register, rather
than just those defined (such as 1300). Any help would be appreciated.

On Mon, Jun 27, 2011 at 4:11 PM, Kurtis Heimerl
<kheimerl at cs.berkeley.edu> wrote:
> Anyhow, here are those three config files:
>
> internal.xml : http://pastebin.freeswitch.org/16609
> acl.conf.xml : http://pastebin.freeswitch.org/16610
> 1300.xml : http://pastebin.freeswitch.org/16611
>
> If anything else could help, I'd love to share it.
>
> The basic story, so far as I see, is that I allow specific IPs through
> the ACL. Somehow this is allowing ANY SIP username to register, rather
> than just those defined (such as 1300). Any help would be appreciated.
>
> On Mon, Jun 27, 2011 at 1:30 PM, Kurtis Heimerl
> <kheimerl at cs.berkeley.edu> wrote:
>> It's enabled in the acl.conf.xml file, using CIDR.
>>
>> What conf files do you consider relevant? acl.conf.xml, internal.xml,
>> a profile or two, anything else?
>>
>> On Mon, Jun 27, 2011 at 1:26 PM, David Ponzone <david.ponzone at ipeva.fr> wrote:
>>> The interesting question is then: why are you able to register without
>>> password, if this feature is not enabled on the profile...
>>> Perhaps you should recap your config once more, and put the relevant files
>>> on PB.
>>> David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr<http://www.ipeva.fr>  -   www.ipeva-studio.com<http://www.ipeva-studio.com>
>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>
>>>
>>>
>>> Le 27/06/2011 ? 20:36, Kurtis Heimerl a ?crit :
>>>
>>> That would explain why removing them didn't do anything!
>>>
>>> Thanks.
>>>
>>> On Mon, Jun 27, 2011 at 6:25 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>>
>>> Just so you know...
>>>
>>>      <param name="accept-blind-reg" value="true"/>
>>>
>>>      <param name="accept-blind-auth" value="true"/>
>>>
>>> These will have no effect in the user directory. They only apply to SIP
>>>
>>> profiles.
>>>
>>> -Steve
>>>
>>>
>>>
>>> On 27 June 2011 02:23, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
>>>
>>> Hello FS Users!
>>>
>>> I'm trying to create the following setup. When a user registers, if
>>>
>>> they register on a known account (lets say X), they do not need a
>>>
>>> password. X's registration is immediately OK'd, and everything is
>>>
>>> great. I've gotten that working using the ACL. The IP address of our
>>>
>>> SIP clients are added through cidr and the clients do not need to give
>>>
>>> passwords.
>>>
>>> However, for some reason, if another account that does not exist in
>>>
>>> the directory (let's say Y) registers, FS returns with a 200 OK,
>>>
>>> instead of rejecting Y. I'm trying to figure out why this is the case,
>>>
>>> and how to remedy that fact.
>>>
>>> I have the following line in my internal.xml file, which I had assumed
>>>
>>> would force this function:
>>>
>>>   <!-- Force the user and auth-user to match. -->
>>>
>>>   <param name="inbound-reg-force-matching-username" value="true"/>
>>>
>>> However, it does not work. In my directory, each individual account as
>>>
>>> the following lines:
>>>
>>>  <user id="1303">
>>>
>>>    <params>
>>>
>>>      <param name="accept-blind-reg" value="true"/>
>>>
>>>      <param name="accept-blind-auth" value="true"/>
>>>
>>>      <param name="vm-password" value="1000"/>
>>>
>>>    </params>
>>>
>>> Though I've found that removing it (from all users in the directory)
>>>
>>> doesn't help.
>>>
>>> I'm primarily concerned with the line in internal.xml; it seems
>>>
>>> possible that the fact that we do not have an auth-user (because we do
>>>
>>> not require auth) means that this won't work. However, I have yet to
>>>
>>> test that hypothesis. The ACL has been the most confusing aspect of
>>>
>>> this installation, with a lot of undocumented aspects, and I get the
>>>
>>> nagging feeling this is another. I could very well be wrong though.
>>>
>>> Thanks for any direction.
>>>
>>> _______________________________________________
>>>
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>>
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>

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

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

!DSPAM:4e090ed032762115621071!



From steveayre at gmail.com  Tue Jun 28 11:56:53 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 28 Jun 2011 08:56:53 +0100
Subject: [Freeswitch-users] Reject SIP registrations
In-Reply-To: <BANLkTikaVD+3Xi_z4SxGrz06GzktmrfHbw@mail.gmail.com>
References: <BANLkTikU6fRYwmUTxzsSHyWs8raukLx4jw@mail.gmail.com>
	<BANLkTimq4YjhjX0vgCt3fsmBcAZRC1OZ-g@mail.gmail.com>
	<BANLkTikQoYWjH8Nsp20Lg4wsy2O0A3L=4Q@mail.gmail.com>
	<53E1375B-A8D6-4C1A-A1DD-055A3527673D@ipeva.fr>
	<BANLkTinrXFE4NYFM4EC7nZqa7jhT-FSLZg@mail.gmail.com>
	<BANLkTinoOwUsjNPSrdXUB4pMkq3LXU1v7w@mail.gmail.com>
	<BANLkTikaVD+3Xi_z4SxGrz06GzktmrfHbw@mail.gmail.com>
Message-ID: <BANLkTiks3KkhqaS6bJLp84D=ve4c-919fA@mail.gmail.com>

When you use a CIDR it matches the user entry based on IP not on username.

You're able to authenticate with other usernames because they're all
authenticating to the same user based on IP.

-Steve



On 28 June 2011 00:12, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:

> One of those links got screwed up...
>
> Anyhow, here are those three config files:
>
> internal.xml : http://bpastebin.freeswitch.org/16609<http://pastebin.freeswitch.org/16609>
>
> acl.conf.xml : http://pastebin.freeswitch.org/16610
>
> 1300.xml : http://pastebin.freeswitch.org/16611
>
> If anything else could help, I'd love to share it.
>
> The basic story, so far as I see, is that I allow specific IPs through
> the ACL. Somehow this is allowing ANY SIP username to register, rather
> than just those defined (such as 1300). Any help would be appreciated.
>
> On Mon, Jun 27, 2011 at 4:11 PM, Kurtis Heimerl
> <kheimerl at cs.berkeley.edu> wrote:
> > Anyhow, here are those three config files:
> >
> > internal.xml : http://pastebin.freeswitch.org/16609
> > acl.conf.xml : http://pastebin.freeswitch.org/16610
> > 1300.xml : http://pastebin.freeswitch.org/16611
> >
> > If anything else could help, I'd love to share it.
> >
> > The basic story, so far as I see, is that I allow specific IPs through
> > the ACL. Somehow this is allowing ANY SIP username to register, rather
> > than just those defined (such as 1300). Any help would be appreciated.
> >
> > On Mon, Jun 27, 2011 at 1:30 PM, Kurtis Heimerl
> > <kheimerl at cs.berkeley.edu> wrote:
> >> It's enabled in the acl.conf.xml file, using CIDR.
> >>
> >> What conf files do you consider relevant? acl.conf.xml, internal.xml,
> >> a profile or two, anything else?
> >>
> >> On Mon, Jun 27, 2011 at 1:26 PM, David Ponzone <david.ponzone at ipeva.fr>
> wrote:
> >>> The interesting question is then: why are you able to register without
> >>> password, if this feature is not enabled on the profile...
> >>> Perhaps you should recap your config once more, and put the relevant
> files
> >>> on PB.
> >>> David Ponzone  Direction Technique
> >>> email: david.ponzone at ipeva.fr
> >>> tel:      01 74 03 18 97
> >>> gsm:   06 66 98 76 34
> >>> Service Client IPeva
> >>> tel:      0811 46 26 26
> >>> www.ipeva.fr  -   www.ipeva-studio.com
> >>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> >>> l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion
> >>> non autoris?e est interdite. Tout message ?lectronique est susceptible
> >>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
> s'il
> >>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de
> ce
> >>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> >>>
> >>>
> >>>
> >>> Le 27/06/2011 ? 20:36, Kurtis Heimerl a ?crit :
> >>>
> >>> That would explain why removing them didn't do anything!
> >>>
> >>> Thanks.
> >>>
> >>> On Mon, Jun 27, 2011 at 6:25 AM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >>>
> >>> Just so you know...
> >>>
> >>>      <param name="accept-blind-reg" value="true"/>
> >>>
> >>>      <param name="accept-blind-auth" value="true"/>
> >>>
> >>> These will have no effect in the user directory. They only apply to SIP
> >>>
> >>> profiles.
> >>>
> >>> -Steve
> >>>
> >>>
> >>>
> >>> On 27 June 2011 02:23, Kurtis Heimerl <kheimerl at cs.berkeley.edu>
> wrote:
> >>>
> >>> Hello FS Users!
> >>>
> >>> I'm trying to create the following setup. When a user registers, if
> >>>
> >>> they register on a known account (lets say X), they do not need a
> >>>
> >>> password. X's registration is immediately OK'd, and everything is
> >>>
> >>> great. I've gotten that working using the ACL. The IP address of our
> >>>
> >>> SIP clients are added through cidr and the clients do not need to give
> >>>
> >>> passwords.
> >>>
> >>> However, for some reason, if another account that does not exist in
> >>>
> >>> the directory (let's say Y) registers, FS returns with a 200 OK,
> >>>
> >>> instead of rejecting Y. I'm trying to figure out why this is the case,
> >>>
> >>> and how to remedy that fact.
> >>>
> >>> I have the following line in my internal.xml file, which I had assumed
> >>>
> >>> would force this function:
> >>>
> >>>   <!-- Force the user and auth-user to match. -->
> >>>
> >>>   <param name="inbound-reg-force-matching-username" value="true"/>
> >>>
> >>> However, it does not work. In my directory, each individual account as
> >>>
> >>> the following lines:
> >>>
> >>>  <user id="1303">
> >>>
> >>>    <params>
> >>>
> >>>      <param name="accept-blind-reg" value="true"/>
> >>>
> >>>      <param name="accept-blind-auth" value="true"/>
> >>>
> >>>      <param name="vm-password" value="1000"/>
> >>>
> >>>    </params>
> >>>
> >>> Though I've found that removing it (from all users in the directory)
> >>>
> >>> doesn't help.
> >>>
> >>> I'm primarily concerned with the line in internal.xml; it seems
> >>>
> >>> possible that the fact that we do not have an auth-user (because we do
> >>>
> >>> not require auth) means that this won't work. However, I have yet to
> >>>
> >>> test that hypothesis. The ACL has been the most confusing aspect of
> >>>
> >>> this installation, with a lot of undocumented aspects, and I get the
> >>>
> >>> nagging feeling this is another. I could very well be wrong though.
> >>>
> >>> Thanks for any direction.
> >>>
> >>> _______________________________________________
> >>>
> >>> Join us at ClueCon 2011, Aug 9-11, Chicago
> >>>
> >>> http://www.cluecon.com 877-7-4ACLUE
> >>>
> >>> FreeSWITCH-users mailing list
> >>>
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>>
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>> _______________________________________________
> >>>
> >>> Join us at ClueCon 2011, Aug 9-11, Chicago
> >>>
> >>> http://www.cluecon.com 877-7-4ACLUE
> >>>
> >>> FreeSWITCH-users mailing list
> >>>
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>>
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>>
> >>> _______________________________________________
> >>> Join us at ClueCon 2011, Aug 9-11, Chicago
> >>> http://www.cluecon.com 877-7-4ACLUE
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>> _______________________________________________
> >>> Join us at ClueCon 2011, Aug 9-11, Chicago
> >>> http://www.cluecon.com 877-7-4ACLUE
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>
> >
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Tue Jun 28 11:58:10 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 28 Jun 2011 08:58:10 +0100
Subject: [Freeswitch-users] .wav vs .gsm file sizes for recording calls.
In-Reply-To: <3C36C8D8-79C7-46F8-BB09-E9321E8B5425@ipeva.fr>
References: <4E0500EE.9040506@499x.com>
	<BANLkTimxdvjSU=gDXMLYLb17jDwamWDSbw@mail.gmail.com>
	<4E08DD6A.5040306@499x.com>
	<A9469BBE-0F9B-4889-B3BD-BCC96487E985@gmail.com>
	<3C36C8D8-79C7-46F8-BB09-E9321E8B5425@ipeva.fr>
Message-ID: <BANLkTimP=TdSz9u2mRFKW9Ju5YxPMgs-bw@mail.gmail.com>

You're right I think. :) Was talking about playback... guess it was pretty
late.

-Steve



On 28 June 2011 00:28, David Ponzone <david.ponzone at ipeva.fr> wrote:

> Steven,
>
> AFAIR, one of the recording app (not sure which) does not support
> mod_native_file, meaning for instance you can't dump your call using codec
> XXX to a raw file.
> That's a quite old limitation, so perhaps it's not an issue anymore.
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 28/06/2011 ? 00:10, Steven Ayre a ?crit :
>
> Its not just voice quality. It's also CPU.
>
> If you're using a compressed format, you're doing transcoding step. FS
> needs to decompress the file to L16 so it can compress it to the codec.
> Storing as wav containing l16 would avoid that. You can also use
> mod_native_file to compress the audio file to each codec format so that you
> don't need to transcode at all at the time because it's already done,
> although that does mean a file for each codec so more space needed.
>
> Steve on iPhone
>
> On 27 Jun 2011, at 20:43, Wes <wes-fs at 499x.com> wrote:
>
> I guess part of my confusion here was due to the term "raw data" mentioned
> in conjunction with the .gsm extension on the wiki page below... but
> actually gsm is a compressed format.
>
>  <http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session
>
> So, what is the best "compressed" format to use for recording voice (that
> is available as a direct recording format inside freeswitch)? There are tons
> of formats listed when I do "show file", but I tried a few and they are also
> giving me large files like the wav extension did. (au, for example)
>
> Even though the PCM/Wave format is preferred for voice quality, when we're
> talking about a 10:1 compression ratio, if the sound quality is still
> acceptable, I'd rather just record directly into the compressed format.
> We're talking about ~10- 20 minute recordings that will need to be
> transferred over the internet to a third party.
>
> On 6/24/2011 6:31 PM, Michael Collins wrote:
>
> I would caution you to consider adding disk space before you try to
> compress all your recordings. The 16 bit SLIN that FS normally puts in your
> wave files are pretty easy to handle, whether playing back in a FS session,
> or encoding for playback on some other device.
>
> An alternative might be to use lame to convert them to MP3's or ogg/vorbis
> files. If you look on the main FS conf call page you'll see I have the
> weekly recordings in multiple formats. (<http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls>
> http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls)
>
> Here are some stats for last Wednesday's call. Note that I record wave
> files in 48kHz then use sox to downsample to 16kHz wave, then I convert that
> 16kHz file into MP3 and Vorbis (in an ogg container). Here's what the
> results look like:
>
> <2831>:ls -1s conf_call_2011-06-15.*
>  18736 conf_call_2011-06-15.mp3
>  23044 conf_call_2011-06-15.ogg
> 199756 conf_call_2011-06-15.wav
>
> <2832>:file conf_call_2011-06-15.mp3
> conf_call_2011-06-15.mp3: MPEG ADTS, layer III, v2,  24 kBits, 16 kHz,
> Monaural
>
> <2833>:file conf_call_2011-06-15.ogg
> conf_call_2011-06-15.ogg: Ogg data, Vorbis audio, mono, 16000 Hz, ~48000
> bps, created by: Xiph.Org libVorbis I
>
> <2834>:file conf_call_2011-06-15.wav
> conf_call_2011-06-15.wav: RIFF (little-endian) data, WAVE audio, Microsoft
> PCM, 16 bit, mono 16000 Hz
>
> Note that the file sizes are in 1K blocks.
>
> So, bottom line is this: if you have the disk space then use wave. If you
> don't have disk space for wave then get some! :D If you REALLY need to use a
> different format then choose something like MP3 or Vorbis for long-term
> storage.
>
> -MC
>
> On Fri, Jun 24, 2011 at 2:26 PM, Wes < <wes-fs at 499x.com>wes-fs at 499x.com>wrote:
>
>> In my tests, if I record a call in .wav format, a 10 second file is
>> about 177,000 bytes, while a 10 second .gsm file is 17,000 bytes.
>>
>> I  then used sox to convert the .gsm file to a .wav file, and it stayed
>> at around 17,000 bytes.  So, is the default recording format for .wav
>> using a higher sample rate? vs the default conversion format for the sox
>> tool?
>>
>> checking the file type using "file" I see that the larger one is:
>> RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
>>
>> and the wav created by sox via the default conversion from .gsm is:
>> RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz
>>
>> So apparently the larger wav file is 16 bit... how are these recording
>> parameters controlled?  Can I set it to record directly into the smaller
>> wav format? Or will I have to run sox on every file...
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>  <http://www.cluecon.com/>http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>>  <FreeSWITCH-users at lists.freeswitch.org>
>> FreeSWITCH-users at lists.freeswitch.org
>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  <http://www.freeswitch.org/>http://www.freeswitch.org
>>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago <http://www.cluecon.com/>http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list <FreeSWITCH-users at lists.freeswitch.org>FreeSWITCH-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE: <http://lists.freeswitch.org/mailman/options/freeswitch-users>http://lists.freeswitch.org/mailman/options/freeswitch-users <http://www.freeswitch.org/>http://www.freeswitch.org
>
>  _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From steveayre at gmail.com  Tue Jun 28 12:00:39 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 28 Jun 2011 09:00:39 +0100
Subject: [Freeswitch-users] does FreeSWITCH care about beeing connected
 to an IPv4- or IPv6-ISP ?
In-Reply-To: <BANLkTikb0DQE8cKxBn_kPuzH47QjBmCXQg@mail.gmail.com>
References: <BANLkTikb0DQE8cKxBn_kPuzH47QjBmCXQg@mail.gmail.com>
Message-ID: <BANLkTin-07-95P3BEh5K1nnB=2FSkvG_cw@mail.gmail.com>

You can use SIP on FS on IPv4, IPv6, or both. You can even bridge calls
inbetween the two (as long as you avoid bypass_media).

When you create a SIP profile it binds to a specific IP, you just need to
put the relevant IPv4/IPv6 IP there. Then treat the profiles as normal. ACLs
also support IPv6 addresses.

Not sure about what the support is in other modules such as mod_h323

-Steve



On 27 June 2011 17:15, bill evergreen <bill.evergreen at gmail.com> wrote:

> Hello,
>
> I intend to setup a little FreeSWITCH test-environment.  Does FreeSWITCH
> care about beeing connected to an IPv4- OR IPv6-isp (it should be possible
> to operate the same environment under IPv4 OR IPv6) ?
> ong
>
> Thank's a lot for any feedback!
>
> Bill
>
>
>
> PS
> I am new to FreeSWITCH :-)
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
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From kheimerl at cs.berkeley.edu  Tue Jun 28 12:34:02 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Tue, 28 Jun 2011 01:34:02 -0700
Subject: [Freeswitch-users] Reject SIP registrations
In-Reply-To: <BANLkTiks3KkhqaS6bJLp84D=ve4c-919fA@mail.gmail.com>
References: <BANLkTikU6fRYwmUTxzsSHyWs8raukLx4jw@mail.gmail.com>
	<BANLkTimq4YjhjX0vgCt3fsmBcAZRC1OZ-g@mail.gmail.com>
	<BANLkTikQoYWjH8Nsp20Lg4wsy2O0A3L=4Q@mail.gmail.com>
	<53E1375B-A8D6-4C1A-A1DD-055A3527673D@ipeva.fr>
	<BANLkTinrXFE4NYFM4EC7nZqa7jhT-FSLZg@mail.gmail.com>
	<BANLkTinoOwUsjNPSrdXUB4pMkq3LXU1v7w@mail.gmail.com>
	<BANLkTikaVD+3Xi_z4SxGrz06GzktmrfHbw@mail.gmail.com>
	<BANLkTiks3KkhqaS6bJLp84D=ve4c-919fA@mail.gmail.com>
Message-ID: <BANLkTimW2KpDGKqAfBvTr4=ZayXvK=qjXw@mail.gmail.com>

Ah, so it's as I worried.

How do I go about creating the situation that I want: users with
accounts are able to authenticate without passwords, but any attempts
to authenticate other accounts (that don't exist) are rejected?

On Tue, Jun 28, 2011 at 12:56 AM, Steven Ayre <steveayre at gmail.com> wrote:
> When you use a CIDR it matches the user entry based on IP not on username.
>
> You're able to authenticate with other usernames because they're all
> authenticating to the same user based on IP.
>
> -Steve
>
>
>
> On 28 June 2011 00:12, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
>>
>> One of those links got screwed up...
>>
>> Anyhow, here are those three config files:
>>
>> internal.xml : http://bpastebin.freeswitch.org/16609
>>
>> acl.conf.xml : http://pastebin.freeswitch.org/16610
>>
>> 1300.xml : http://pastebin.freeswitch.org/16611
>>
>> If anything else could help, I'd love to share it.
>>
>> The basic story, so far as I see, is that I allow specific IPs through
>> the ACL. Somehow this is allowing ANY SIP username to register, rather
>> than just those defined (such as 1300). Any help would be appreciated.
>>
>> On Mon, Jun 27, 2011 at 4:11 PM, Kurtis Heimerl
>> <kheimerl at cs.berkeley.edu> wrote:
>> > Anyhow, here are those three config files:
>> >
>> > internal.xml : http://pastebin.freeswitch.org/16609
>> > acl.conf.xml : http://pastebin.freeswitch.org/16610
>> > 1300.xml : http://pastebin.freeswitch.org/16611
>> >
>> > If anything else could help, I'd love to share it.
>> >
>> > The basic story, so far as I see, is that I allow specific IPs through
>> > the ACL. Somehow this is allowing ANY SIP username to register, rather
>> > than just those defined (such as 1300). Any help would be appreciated.
>> >
>> > On Mon, Jun 27, 2011 at 1:30 PM, Kurtis Heimerl
>> > <kheimerl at cs.berkeley.edu> wrote:
>> >> It's enabled in the acl.conf.xml file, using CIDR.
>> >>
>> >> What conf files do you consider relevant? acl.conf.xml, internal.xml,
>> >> a profile or two, anything else?
>> >>
>> >> On Mon, Jun 27, 2011 at 1:26 PM, David Ponzone <david.ponzone at ipeva.fr>
>> >> wrote:
>> >>> The interesting question is then: why are you able to register without
>> >>> password, if this feature is not enabled on the profile...
>> >>> Perhaps you should recap your config once more, and put the relevant
>> >>> files
>> >>> on PB.
>> >>> David Ponzone ?Direction Technique
>> >>> email: david.ponzone at ipeva.fr
>> >>> tel: ? ? ?01 74 03 18 97
>> >>> gsm: ? 06 66 98 76 34
>> >>> Service Client?IPeva
>> >>> tel: ? ? ?0811 46 26 26
>> >>> www.ipeva.fr? -? ?www.ipeva-studio.com
>> >>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>> >>> ?
>> >>> l'intention exclusive de ses destinataires. Toute utilisation ou
>> >>> diffusion
>> >>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> >>> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce
>> >>> message s'il
>> >>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de
>> >>> ce
>> >>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>> >>>
>> >>>
>> >>>
>> >>> Le 27/06/2011 ? 20:36, Kurtis Heimerl a ?crit :
>> >>>
>> >>> That would explain why removing them didn't do anything!
>> >>>
>> >>> Thanks.
>> >>>
>> >>> On Mon, Jun 27, 2011 at 6:25 AM, Steven Ayre <steveayre at gmail.com>
>> >>> wrote:
>> >>>
>> >>> Just so you know...
>> >>>
>> >>> ? ?? <param name="accept-blind-reg" value="true"/>
>> >>>
>> >>> ? ? ?<param name="accept-blind-auth" value="true"/>
>> >>>
>> >>> These will have no effect in the user directory. They only apply to
>> >>> SIP
>> >>>
>> >>> profiles.
>> >>>
>> >>> -Steve
>> >>>
>> >>>
>> >>>
>> >>> On 27 June 2011 02:23, Kurtis Heimerl <kheimerl at cs.berkeley.edu>
>> >>> wrote:
>> >>>
>> >>> Hello FS Users!
>> >>>
>> >>> I'm trying to create the following setup. When a user registers, if
>> >>>
>> >>> they register on a known account (lets say X), they do not need a
>> >>>
>> >>> password. X's registration is immediately OK'd, and everything is
>> >>>
>> >>> great. I've gotten that working using the ACL. The IP address of our
>> >>>
>> >>> SIP clients are added through cidr and the clients do not need to give
>> >>>
>> >>> passwords.
>> >>>
>> >>> However, for some reason, if another account that does not exist in
>> >>>
>> >>> the directory (let's say Y) registers, FS returns with a 200 OK,
>> >>>
>> >>> instead of rejecting Y. I'm trying to figure out why this is the case,
>> >>>
>> >>> and how to remedy that fact.
>> >>>
>> >>> I have the following line in my internal.xml file, which I had assumed
>> >>>
>> >>> would force this function:
>> >>>
>> >>> ? <!-- Force the user and auth-user to match. -->
>> >>>
>> >>> ? <param name="inbound-reg-force-matching-username" value="true"/>
>> >>>
>> >>> However, it does not work. In my directory, each individual account as
>> >>>
>> >>> the following lines:
>> >>>
>> >>> ?<user id="1303">
>> >>>
>> >>> ? ?<params>
>> >>>
>> >>> ? ? ?<param name="accept-blind-reg" value="true"/>
>> >>>
>> >>> ? ? ?<param name="accept-blind-auth" value="true"/>
>> >>>
>> >>> ? ? ?<param name="vm-password" value="1000"/>
>> >>>
>> >>> ? ?</params>
>> >>>
>> >>> Though I've found that removing it (from all users in the directory)
>> >>>
>> >>> doesn't help.
>> >>>
>> >>> I'm primarily concerned with the line in internal.xml; it seems
>> >>>
>> >>> possible that the fact that we do not have an auth-user (because we do
>> >>>
>> >>> not require auth) means that this won't work. However, I have yet to
>> >>>
>> >>> test that hypothesis. The ACL has been the most confusing aspect of
>> >>>
>> >>> this installation, with a lot of undocumented aspects, and I get the
>> >>>
>> >>> nagging feeling this is another. I could very well be wrong though.
>> >>>
>> >>> Thanks for any direction.
>> >>>
>> >>> _______________________________________________
>> >>>
>> >>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> >>>
>> >>> http://www.cluecon.com 877-7-4ACLUE
>> >>>
>> >>> FreeSWITCH-users mailing list
>> >>>
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>>
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>
>> >>> http://www.freeswitch.org
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>>
>> >>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> >>>
>> >>> http://www.cluecon.com 877-7-4ACLUE
>> >>>
>> >>> FreeSWITCH-users mailing list
>> >>>
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>>
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>
>> >>> http://www.freeswitch.org
>> >>>
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> >>> http://www.cluecon.com 877-7-4ACLUE
>> >>>
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> >>> http://www.cluecon.com 877-7-4ACLUE
>> >>>
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>>
>> >>
>> >
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
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From kris at livecall.com  Tue Jun 28 13:49:19 2011
From: kris at livecall.com (Kris)
Date: Tue, 28 Jun 2011 02:49:19 -0700
Subject: [Freeswitch-users] Phrase macros in conferences
References: <FE02E1C7-64EA-4E85-AC33-B491AB2FF555@bryansmart.com><BANLkTikRKyF8eppAtqd6DbQqRH29JLnQ7w@mail.gmail.com><93AE9C49-C6F6-44E6-BC7E-5A31AC659C4C@bryansmart.com><9CE9932A-AD27-4183-9C2E-75148116A23B@tech21.com>
	<22CBC366-3CD3-4429-80B6-961AF1566D1C@bryansmart.com>
Message-ID: <B8CF9984153543B69B76337FDDD3BA3F@stor1>

The conference controls/profiles cant be changed after the conference 
starts. Do You know C? We all have to do some C when Freeswitch can't do 
what we want.A feature would have to be added to mod_conference to play 
files unpon joining the conference. It would have to be something like this:
<action application="conference" 
data="${forum_ext}-${domain_name}@${conference_controls},PlayOnEntry=${name_var}&conf-enter.wav,PlayOnExit=???"/>

To really do the fancier stuff, you have to capture the CUSTOM event. Just 
do it with 2 API calls. Both are scheduled to play one after the other.
Api.Execute("conference", ConferenceName + " play " + "name.wav"); 
//+MemberID);

Api.Execute("conference", ConferenceName + " play " + "conf-enter.wav"); 
//+MemberID);

----- Original Message ----- 
From: "Bryan Smart" <bryansmart at bryansmart.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, June 27, 2011 1:03 PM
Subject: Re: [Freeswitch-users] Phrase macros in conferences


Yes, but not sure how I'd do that. In the conference profile, I can play a 
single sound file for an enter/leave. I can't join two files with "!" to 
have them play in series, like "name.wav!conference/conf-enter.wav".

Bryan

On Jun 27, 2011, at 3:37 PM, Brad Mina wrote:

You could have users prompted to record their name to get around the first 
issue and play it back before "... Has joined the conference" is played 
back.

Sent from my iPhone

On Jun 27, 2011, at 12:05 PM, Bryan Smart 
<bryansmart at bryansmart.com<mailto:bryansmart at bryansmart.com>> wrote:

Thanks, Michael. At least I know that it is a limitation, and not me 
overlooking something. :) I have 3 or 4 years on Asterisk, but only months 
on Freeswitch, so I'm always concerned that I haven't dug enough before 
posting requests.

For phrases inside conferences, I imagine 3 cases.

First are dynamic announcements, such as those that announce when someone 
joins or leaves a conference. It is helpful if the joining or leaving party 
can be identified. I imagined combining TTS for the name, along with a 
pre-recorded fragment like conference/conf-has_joined.wav. The result would 
be like "John has entered the conference." Right now, I must use entirely 
TTS for such announcements. I can't use "!" to concatenate files, and I 
can't defer to a phrase macro to string TTS and recordings together.

Second applies to localization. In a dynamic announcement, it might not be 
appropriate for the noun to be spoken first in languages other than English. 
Right now, the conference's prompt language must be customized by creating a 
different conference profile that specifies the correct sound base for the 
language. That could mean that the number of profiles in use has to be 
multiplied by the number of languages that should be supported. I know that 
the profile can omit the sound base, and the sound base of the first person 
to join will be used, but, particularly in a situation that concatenates TTS 
and audio, that can be worse. If the first person to join automatically 
selects sound files for a language, but the concatenation order is different 
for that language, then any rules in that profile that concatenate TTS and 
recordings to render announcements could result in strange outcomes. As far 
as I've been able to discover, the phrase system is the appropriate 
Freeswitch way to localize announcements, so it would probably be best for 
Conference to work that way also, rather than depending on per-case 
localization being handled with multiple profiles. If phrases were 
supported, then a single conference profile could play both static and 
dynamic prompts, and could automatically adjust to any language where phrase 
macros are present on the system.

Last are cases when multiple types of notification are desired. For example, 
notifications might use a combination of cues and speech. An audio cue 
serves a similar function to a visual icon: attracts attention and conveys 
general meaning. In a situation like a conference, there might be soft audio 
cues for joining, leaving, when a new person has the floor, etc. When one of 
those events takes place, the cue plays first to attract attention and set 
the listener's expectations regarding the type of spoken information that 
will follow. Next, an announcement is heard with details about the event. 
"<cue> John has left  the conference."






On Jun 27, 2011, at 11:25 AM, Michael Collins wrote:

Phrase macros are not supported inside mod_conference. If you look inside 
mod_conference.c you will see that it has its own functions for injecting 
audio into the conference and it does not support phrase macros. You can 
open up a feature request on Jira. I honestly don't know how easy/difficult 
this would be, so you may consider offering a bounty to sweeten the deal.

Curious: what types of phrases are you trying to play into the conference?

-MC

On Sun, Jun 26, 2011 at 4:57 PM, Bryan Smart 
<<mailto:bryansmart at bryansmart.com>bryansmart at bryansmart.com<mailto:bryansmart at bryansmart.com>> 
wrote:
I'm attempting to use phrase macros in mod_conference, but I'm not having 
success. Perhaps they aren't supported?

I have a working phrase macro. In the dialplan, both

<action application="phrase" data="my_phrase"/>

and

<action application="playback" data="phrase:my_phrase"/>

Work just fine.

The wiki says that, in conference.conf.xml, entries like pin-sound accept, 
for their value, file paths, strings prefixed with "say" for TTS, and 
strings prefixed with "tone_stream". This looks almost the same as what can 
be passed to the playback application in the dialplan. When I try to prefix 
a string with "phrase", it doesn't work.

<param name="pin-sound" value="phrase:my_phrase"/>

The console tells me "unknown file format". The console still tells me this 
if I make up phrase names

<param name="pin-sound" value="phrase:not_a_real_phrase"/>

So it is complaining about the syntax or capability, rather than an 
unrecognized format of an actual file.

If mod_conference doesn't support phrase macros, is this a near-future 
feature? Phrases would seem to be necessary to ocalize announcements and 
play announcements that are stitched together from multiple files.

Bryan


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From freeswitch at ml102.pinguin.uni.cc  Tue Jun 28 14:42:50 2011
From: freeswitch at ml102.pinguin.uni.cc (Al Bogner)
Date: Tue, 28 Jun 2011 12:42:50 +0200
Subject: [Freeswitch-users] SIP/2.0 403 Forbidden
In-Reply-To: <77CE8968-A79D-4310-964D-E4952F1D436C@ipeva.fr>
References: <20110627220504.377cc403@ml102.pinguin.uni.cc>
	<77CE8968-A79D-4310-964D-E4952F1D436C@ipeva.fr>
Message-ID: <20110628124250.688df723@ml102.pinguin.uni.cc>

Am Di, 28 Jun 2011 01:55:08 CEST schrieb David Ponzone:

Hi David,
 
> please, show us the content of the following files:
> conf/directory/default.xml
> conf/vars.xml

These files are default.

See below.

> May you also tell us if you see a message in fs_cli when you try to
> register ?

/opt/freeswitch/bin/fs_cli
Type /help <enter> to see a list of commands
+OK log level  [7]

I see nothing else. Should I use a specific option? Do you want me to
try another sipclient than twinkle? Linphone is installed too.

Al


cat /opt/freeswitch/conf/directory/default.xml 
<!--
    NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE
NOTICE 
    FreeSWITCH works off the concept of users and domains just like
email. You have users that are in domains for example 1000 at domain.com.
    
    When freeswitch gets a register packet it looks for the user in the
directory based on the from or to domain in the packet depending on how
your sofia profile is configured.  Out of the box the default domain
will be the IP address of the machine running FreeSWITCH.  This IP can
be found by typing "sofia status" at the CLI.  You will register your
phones to the IP and not the hostname by default. If you wish to
register using the domain please open vars.xml in the root conf
directory and set the default domain to the hostname you desire.  Then
you would use the domain name in the client instead of the IP address
to register with FreeSWITCH. 
    NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE
NOTICE -->

<include>
  <!--the domain or ip (the right hand side of the @ in the addr-->
  <domain name="$${domain}">
    <params>
      <param name="dial-string"
value="{sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
</params>

    <variables>
      <variable name="record_stereo" value="true"/>
      <variable name="default_gateway" value="$${default_provider}"/>
      <variable name="default_areacode" value="$${default_areacode}"/>
      <variable name="transfer_fallback_extension" value="operator"/>
    </variables>

    <groups>
      <group name="default">
	<users>
	  <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
	</users>
      </group>

      <group name="sales">
	<users>
	  <!--
	      type="pointer" is a pointer so you can have the 
	      same user in multiple groups.  It basically means
	      to keep searching for the user in the directory.
	  -->
	  <user id="1000" type="pointer"/>
	  <user id="1001" type="pointer"/>
	  <user id="1002" type="pointer"/>
	  <user id="1003" type="pointer"/>
	  <user id="1004" type="pointer"/>
	</users>
      </group>

      <group name="billing">
	<users>
	  <user id="1005" type="pointer"/>
	  <user id="1006" type="pointer"/>
	  <user id="1007" type="pointer"/>
	  <user id="1008" type="pointer"/>
	  <user id="1009" type="pointer"/>
	</users>
      </group>

      <group name="support">
	<users>
	  <user id="1010" type="pointer"/>
	  <user id="1011" type="pointer"/>
	  <user id="1012" type="pointer"/>
	  <user id="1013" type="pointer"/>
	  <user id="1014" type="pointer"/>
	</users>
      </group>
    </groups>

  </domain>
</include>





cat /opt/freeswitch/conf/vars.xml 
<include>
  <!-- Preprocessor Variables
       These are introduced when configuration strings must be
consistent across modules. NOTICE: YOU CAN NOT COMMENT OUT AN
X-PRE-PROCESS line, Remove the line instead. 
       WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
WARNING WARNING 
       YOU SHOULD CHANGE THIS default_password value if you don't want
to be subject to any toll fraud in the future.  It's your
responsibility to secure your own system. 
       This default config is used to demonstrate the feature set of
FreeSWITCH. 
       WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
WARNING WARNING -->
  <X-PRE-PROCESS cmd="set" data="default_password=1234"/>
  <!-- Did you change it yet? -->

  <X-PRE-PROCESS cmd="set"
  data="sound_prefix=$${sounds_dir}/en/us/callie"/>

  <!--
      This setting is what sets the default domain FreeSWITCH will use
  if all else fails. 
      FreeSWICH will default to $${local_ip_v4} unless changed.
  Changing this setting does affect the sip authentication.  Please
  review conf/directory/default.xml for more information on this topic.
  -->
  <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
  <X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/>
  <X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/>
  <X-PRE-PROCESS cmd="set" data="use_profile=internal"/>

  <!--
      Enable ZRTP globally you can override this on a per channel basis
      
      http://wiki.freeswitch.org/wiki/ZRTP (on how to enable zrtp)
  -->
  <X-PRE-PROCESS cmd="set" data="zrtp_secure_media=true"/>

  <!-- 
       Examples of codec options: (module must be compiled and loaded)
       
       codecname[@8000h|16000h|32000h[@XXi]]
       
       XX is the frame size must be multples allowed for the codec
       FreeSWITCH can support 10-120ms on some codecs. 
       We do not support exceeding the MTU of the RTP packet.


       iLBC at 30i         - iLBC using mode=30 which will win in all
       cases. DVI4 at 8000h@20i   - IMA ADPCM 8kHz using 20ms ptime.
       (multiples of 10) DVI4 at 16000h@40i  - IMA ADPCM 16kHz using 40ms
       ptime. (multiples of 10) speex at 8000h@20i  - Speex 8kHz using
       20ms ptime. speex at 16000h@20i - Speex 16kHz using 20ms ptime.
       speex at 32000h@20i - Speex 32kHz using 20ms ptime.
       BV16             - BroadVoice 16kb/s narrowband, 8kHz
       BV32             - BroadVoice 32kb/s wideband, 16kHz
       G7221 at 16000h     - G722.1 16kHz (aka Siren 7)
       G7221 at 32000h     - G722.1C 32kHz (aka Siren 14)
       CELT at 32000h      - CELT 32kHz, only 10ms supported
       CELT at 48000h      - CELT 48kHz, only 10ms supported
       GSM at 40i          - GSM 8kHz using 40ms ptime. (GSM is done in
       multiples of 20, Default is 20ms) G722             - G722 16kHz
       using default 20ms ptime. (multiples of 10) PCMU             -
       G711 8kHz ulaw using default 20ms ptime. (multiples of 10)
       PCMA             - G711 8kHz alaw using default 20ms ptime.
       (multiples of 10) G726-16          - G726 16kbit adpcm using
       default 20ms ptime. (multiples of 10) G726-24          - G726
       24kbit adpcm using default 20ms ptime. (multiples of 10)
       G726-32          - G726 32kbit adpcm using default 20ms ptime.
       (multiples of 10) G726-40          - G726 40kbit adpcm using
       default 20ms ptime. (multiples of 10) AAL2-G726-16     - Same as
       G726-16 but using AAL2 packing. (multiples of 10)
       AAL2-G726-24     - Same as G726-24 but using AAL2 packing.
       (multiples of 10) AAL2-G726-32     - Same as G726-32 but using
       AAL2 packing. (multiples of 10) AAL2-G726-40     - Same as
       G726-40 but using AAL2 packing. (multiples of 10)
       LPC              - LPC10 using 90ms ptime (only supports 90ms at
       this time in FreeSWITCH) L16              - L16 isn't
       recommended for VoIP but you can do it. L16 can exceed the MTU
       rather quickly. These are the passthru audio codecs:
       G729             - G729 in passthru mode. (mod_g729)
       G723             - G723.1 in passthru mode. (mod_g723_1)
       AMR              - AMR in passthru mode. (mod_amr) These are the
       passthru video codecs: (mod_h26x) H261             - H.261 Video
       H263             - H.263 Video
       H263-1998        - H.263-1998 Video
       H263-2000        - H.263-2000 Video
       H264             - H.264 Video
       
       RTP Dynamic Payload Numbers currently used in FreeSWITCH and
       what for.

       96  - AMR
       97  - iLBC (30)
       98  - iLBC (20)
       99  - Speex 8kHz, 16kHz, 32kHz
       100 -
       101 - telephone-event
       102 -
       103 - 
       104 - 
       105 - 
       106 - BV16
       107 - G722.1 (16kHz)
       108 -
       109 -
       110 -
       111 -
       112 -
       113 -
       114 - CELT 32kHz, 48kHz
       115 - G722.1C (32kHz)
       116 -
       117 - SILK 8kHz
       118 - SILK 12kHz
       119 - SILK 16kHz
       120 - SILK 24kHz
       121 - AAL2-G726-40 && G726-40
       122 - AAL2-G726-32 && G726-32
       123 - AAL2-G726-24 && G726-24
       124 - AAL2-G726-16 && G726-16
       125 - 
       126 -
       127 - BV32

  -->
  <X-PRE-PROCESS cmd="set"
  data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>

  <!--
      xmpp_client_profile and xmpp_server_profile
      xmpp_client_profile can be any string. 
      xmpp_server_profile is appended to "dingaling_" to form the
  database name containing the "subscriptions" table.
      used by: dingaling.conf.xml enum.conf.xml 
  --> 

  <X-PRE-PROCESS cmd="set" data="xmpp_client_profile=xmppc"/>
  <X-PRE-PROCESS cmd="set" data="xmpp_server_profile=xmpps"/>
  <!-- 
       THIS IS ONLY USED FOR DINGALING

       bind_server_ip

       Can be an ip address, a dns name, or "auto". 
       This determines an ip address available on this host to bind.
       If you are separating RTP and SIP traffic, you will want to have
       use different addresses where this variable appears.
       Used by: dingaling.conf.xml
  -->
  <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>

  <!-- NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE
  NOTICE NOTICE 
       If you're going to load test FreeSWITCH please input real IP
  addresses for external_rtp_ip and external_sip_ip
  -->

  <!-- external_rtp_ip
       Can be an one of:
           ip address: "12.34.56.78"
           a stun server lookup: "stun:stun.server.com"
           a DNS name: "host:host.server.com"
       where fs.mydomain.com is a DNS A record-useful when fs is on
       a dynamic IP address, and uses a dynamic DNS updater.
       If unspecified, the bind_server_ip value is used.
       Used by: sofia.conf.xml dingaling.conf.xml
  -->
  <X-PRE-PROCESS cmd="set"
  data="external_rtp_ip=stun:stun.freeswitch.org"/>

  <!-- external_sip_ip
      Used as the public IP address for SDP.
       Can be an one of:
           ip address: "12.34.56.78"
           a stun server lookup: "stun:stun.server.com"
           a DNS name: "host:host.server.com"
       where fs.mydomain.com is a DNS A record-useful when fs is on
       a dynamic IP address, and uses a dynamic DNS updater.
       If unspecified, the bind_server_ip value is used.
       Used by: sofia.conf.xml dingaling.conf.xml
  -->
  <X-PRE-PROCESS cmd="set"
  data="external_sip_ip=stun:stun.freeswitch.org"/>

  <!-- unroll-loops
       Used to turn on sip loopback unrolling.
  --> 
  <X-PRE-PROCESS cmd="set" data="unroll_loops=true"/>

  <!-- outbound_caller_id and outbound_caller_name
       The caller ID telephone number we should use when calling out.
       Used by: conference.conf.xml and user directory for default
       outbound callerid name and number.
  -->
  <X-PRE-PROCESS cmd="set" data="outbound_caller_name=FreeSWITCH"/>
  <X-PRE-PROCESS cmd="set" data="outbound_caller_id=0000000000"/>

  <!-- various debug and defaults -->
  <X-PRE-PROCESS cmd="set" data="call_debug=false"/>
  <X-PRE-PROCESS cmd="set" data="console_loglevel=info"/>
  <X-PRE-PROCESS cmd="set" data="default_areacode=918"/>
  <X-PRE-PROCESS cmd="set" data="default_country=US"/>

  <X-PRE-PROCESS cmd="set" data="be-ring=%(1000,3000,425)"/>
  <X-PRE-PROCESS cmd="set" data="ca-ring=%(2000,4000,440,480)"/>
  <X-PRE-PROCESS cmd="set" data="cn-ring=%(1000,4000,450)"/>
  <X-PRE-PROCESS cmd="set" data="cy-ring=%(1500,3000,425)"/>
  <X-PRE-PROCESS cmd="set" data="cz-ring=%(1000,4000,425)"/>
  <X-PRE-PROCESS cmd="set" data="de-ring=%(1000,4000,425)"/>
  <X-PRE-PROCESS cmd="set" data="dk-ring=%(1000,4000,425)"/>
  <X-PRE-PROCESS cmd="set" data="dz-ring=%(1500,3500,425)"/>
  <X-PRE-PROCESS cmd="set" data="eg-ring=%(2000,1000,475,375)"/>
  <X-PRE-PROCESS cmd="set" data="fi-ring=%(1000,4000,425)"/>
  <X-PRE-PROCESS cmd="set" data="fr-ring=%(1500,3500,440)"/>
  <X-PRE-PROCESS cmd="set"
  data="hk-ring=%(400,200,440,480);%(400,3000,440,480)"/>
  <X-PRE-PROCESS cmd="set" data="hu-ring=%(1250,3750,425)"/>
  <X-PRE-PROCESS cmd="set" data="il-ring=%(1000,3000,400)"/>
  <X-PRE-PROCESS cmd="set"
  data="in-ring=%(400,200,425,375);%(400,2000,425,375)"/>
  <X-PRE-PROCESS cmd="set" data="jp-ring=%(1000,2000,420,380)"/>
  <X-PRE-PROCESS cmd="set" data="ko-ring=%(1000,2000,440,480)"/>
  <X-PRE-PROCESS cmd="set" data="pk-ring=%(1000,2000,400)"/>
  <X-PRE-PROCESS cmd="set" data="pl-ring=%(1000,4000,425)"/>
  <X-PRE-PROCESS cmd="set" data="ro-ring=%(1850,4150,475,425)"/>
  <X-PRE-PROCESS cmd="set" data="rs-ring=%(1000,4000,425)"/>
  <X-PRE-PROCESS cmd="set" data="ru-ring=%(800,3200,425)"/>
  <X-PRE-PROCESS cmd="set" data="sa-ring=%(1200,4600,425)"/>
  <X-PRE-PROCESS cmd="set" data="tr-ring=%(2000,4000,450)"/>
  <X-PRE-PROCESS cmd="set"
  data="uk-ring=%(400,200,400,450);%(400,2000,400,450)"/>
  <X-PRE-PROCESS cmd="set" data="us-ring=%(2000,4000,440,480)"/>
  <X-PRE-PROCESS cmd="set"
  data="bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440)"/>
  <X-PRE-PROCESS cmd="set"
  data="sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)"/> <!--
  Setting up your default sip provider is easy. Below are some values
  that should work in most cases. These are for
  conf/directory/default/example.com.xml --> <X-PRE-PROCESS cmd="set"
  data="default_provider=example.com"/> <X-PRE-PROCESS cmd="set"
  data="default_provider_username=joeuser"/> <X-PRE-PROCESS cmd="set"
  data="default_provider_password=password"/> <X-PRE-PROCESS cmd="set"
  data="default_provider_from_domain=example.com"/> <!-- true or false
  --> <X-PRE-PROCESS cmd="set" data="default_provider_register=false"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_contact=5000"/>

  <!--
      SIP and TLS settings. http://wiki.freeswitch.org/wiki/Tls
  -->
  <X-PRE-PROCESS cmd="set" data="sip_tls_version=tlsv1"/>

  <!-- Internal SIP Profile -->
  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>
  <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/>
  <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/>
  <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/>
  <X-PRE-PROCESS cmd="set"
  data="internal_ssl_dir=$${base_dir}/conf/ssl"/>

  <!-- External SIP Profile -->
  <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>
  <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/>
  <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/>
  <X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/>
  <X-PRE-PROCESS cmd="set"
  data="external_ssl_dir=$${base_dir}/conf/ssl"/> </include>




> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
> ? l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion non autoris?e est interdite. Tout message ?lectronique est
> susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre
> de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes
> pas destinataire de ce message, merci de le d?truire imm?diatement et
> d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 27/06/2011 ? 22:05, Al Bogner a ?crit :
> 
> > I am doing my first steps with freeswitch. 
> > 
> > I have installed on an Ubuntu 11.04 server:
> > 
> > ii  freeswitch                           1.0.7~20110603-0natty6  
> > ii  freeswitch-codec-passthru-g7231      1.0.7~20110603-0natty6  
> > ii  freeswitch-codec-passthru-g729       1.0.7~20110603-0natty6  
> > ii  freeswitch-lang-de                   1.0.7~20110603-0natty6  
> > ii  freeswitch-lang-en                   1.0.7~20110603-0natty6
> > 
> > 
> > Following the tutorial at
> > http://www.onlinesolution.co.nz/viewtopic.php?t=102
> > I modified /opt/freeswitch/conf/directory/default/1000.xml
> > 
> > <include>
> >  <user id="1000">
> >    <params>
> >      <param name="password" value="siptest"/>
> >      <param name="vm-password" value="siptest"/>
> >    </params>
> >    <variables>
> >      <variable name="toll_allow" value="domestic,local"/>
> >      <variable name="accountcode" value="1000"/>
> >      <variable name="user_context" value="default"/>
> >      <variable name="effective_caller_id_name" value="Extension
> > 1000"/> <variable name="effective_caller_id_number" value="1000"/>
> >      <variable name="outbound_caller_id_name"
> > value="$${outbound_caller_name}"/> <variable
> > name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
> > <variable name="callgroup" value="techsupport"/> </variables>
> >  </user>
> > </include>
> > 
> > 
> > Then I tried to connect to the server from another machine:
> > 
> > 
> > +++ 27-6-2011 17:07:30.279796 INFO SIP ::send_sip_udp
> > Send to: udp:192.168.2.100:5060
> > REGISTER sip:192.168.2.100 SIP/2.0
> > Via: SIP/2.0/UDP 192.168.2.108;rport;branch=z9hG4bKcvkohckk
> > Max-Forwards: 70
> > To: "FS Admin" <sip:1000 at 192.168.2.100>
> > From: "FS Admin" <sip:1000 at 192.168.2.100>;tag=nyyvz
> > Call-ID: zmjxppampyrrgxk at client.local.tld
> > CSeq: 645 REGISTER
> > Contact: <sip:1000 at 192.168.2.108>;expires=3600
> > Allow:
> > INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
> > User-Agent: Twinkle/1.4.2 Content-Length: 0
> > 
> > 
> > ---
> > 
> > +++ 27-6-2011 17:07:30.281799 INFO SIP ::process_sip_msg
> > Received from: udp:192.168.2.100:5060
> > SIP/2.0 403 Forbidden
> > Via: SIP/2.0/UDP
> > 192.168.2.108;received=192.168.2.108;rport=5060;branch=z9hG4bKcvkohckk
> > To: "FS Admin" <sip:1000 at 192.168.2.100>;tag=iairy From: "FS Admin"
> > <sip:1000 at 192.168.2.100>;tag=nyyvz Call-ID:
> > zmjxppampyrrgxk at client.local.tld CSeq: 645 REGISTER
> > Server: Twinkle/1.4.2
> > Content-Length: 0
> > 
> > 
> > What could be wrong, so I can't connect?
> > 
> > Al
> > 
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 



From avi at avimarcus.net  Tue Jun 28 15:02:06 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 28 Jun 2011 14:02:06 +0300
Subject: [Freeswitch-users] does FreeSWITCH care about beeing connected
 to an IPv4- or IPv6-ISP ?
In-Reply-To: <BANLkTin-07-95P3BEh5K1nnB=2FSkvG_cw@mail.gmail.com>
References: <BANLkTikb0DQE8cKxBn_kPuzH47QjBmCXQg@mail.gmail.com>
	<BANLkTin-07-95P3BEh5K1nnB=2FSkvG_cw@mail.gmail.com>
Message-ID: <BANLkTinCy3Y5ezcB3-y04+V9TboN4cY=zA@mail.gmail.com>

Just to clarify: each sip profile binds to ONE and only ONE IP address. That
can be ipv4 external, local, ipv6, whatever, but you need a separate profile
for EACH IP you wish to get calls on.

-Avi



On Tue, Jun 28, 2011 at 11:00 AM, Steven Ayre <steveayre at gmail.com> wrote:

> You can use SIP on FS on IPv4, IPv6, or both. You can even bridge calls
> inbetween the two (as long as you avoid bypass_media).
>
> When you create a SIP profile it binds to a specific IP, you just need to
> put the relevant IPv4/IPv6 IP there. Then treat the profiles as normal. ACLs
> also support IPv6 addresses.
>
> Not sure about what the support is in other modules such as mod_h323
>
> -Steve
>
>
>
> On 27 June 2011 17:15, bill evergreen <bill.evergreen at gmail.com> wrote:
>
>> Hello,
>>
>> I intend to setup a little FreeSWITCH test-environment.  Does FreeSWITCH
>> care about beeing connected to an IPv4- OR IPv6-isp (it should be possible
>> to operate the same environment under IPv4 OR IPv6) ?
>> ong
>>
>> Thank's a lot for any feedback!
>>
>> Bill
>>
>>
>>
>> PS
>> I am new to FreeSWITCH :-)
>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Tue Jun 28 15:08:19 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 28 Jun 2011 12:08:19 +0100
Subject: [Freeswitch-users] does FreeSWITCH care about beeing connected
	to an IPv4- or IPv6-ISP ?
In-Reply-To: <BANLkTinCy3Y5ezcB3-y04+V9TboN4cY=zA@mail.gmail.com>
References: <BANLkTikb0DQE8cKxBn_kPuzH47QjBmCXQg@mail.gmail.com>
	<BANLkTin-07-95P3BEh5K1nnB=2FSkvG_cw@mail.gmail.com>
	<BANLkTinCy3Y5ezcB3-y04+V9TboN4cY=zA@mail.gmail.com>
Message-ID: <B444ABD6-E73E-4F5D-8A11-710DAF512773@gmail.com>

Yep, sorry I wasn't clear enough. You can bridge between 4/6 by bridging between different profiles bound to each.

Steve on iPhone

On 28 Jun 2011, at 12:02, Avi Marcus <avi at avimarcus.net> wrote:

> Just to clarify: each sip profile binds to ONE and only ONE IP address. That can be ipv4 external, local, ipv6, whatever, but you need a separate profile for EACH IP you wish to get calls on.
> 
> -Avi
> 
> 
> 
> On Tue, Jun 28, 2011 at 11:00 AM, Steven Ayre <steveayre at gmail.com> wrote:
> You can use SIP on FS on IPv4, IPv6, or both. You can even bridge calls inbetween the two (as long as you avoid bypass_media).
> 
> When you create a SIP profile it binds to a specific IP, you just need to put the relevant IPv4/IPv6 IP there. Then treat the profiles as normal. ACLs also support IPv6 addresses.
> 
> Not sure about what the support is in other modules such as mod_h323
> 
> -Steve
> 
> 
> 
> On 27 June 2011 17:15, bill evergreen <bill.evergreen at gmail.com> wrote:
> Hello,
> 
> I intend to setup a little FreeSWITCH test-environment.  Does FreeSWITCH care about beeing connected to an IPv4- OR IPv6-isp (it should be possible to operate the same environment under IPv4 OR IPv6) ?
> ong
> 
> Thank's a lot for any feedback! 
> 
> Bill
> 
> 
> 
> PS
> I am new to FreeSWITCH :-)
> 
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From david.ponzone at ipeva.fr  Tue Jun 28 15:19:19 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 28 Jun 2011 13:19:19 +0200
Subject: [Freeswitch-users] SIP/2.0 403 Forbidden
In-Reply-To: <20110628124250.688df723@ml102.pinguin.uni.cc>
References: <20110627220504.377cc403@ml102.pinguin.uni.cc>
	<77CE8968-A79D-4310-964D-E4952F1D436C@ipeva.fr>
	<20110628124250.688df723@ml102.pinguin.uni.cc>
Message-ID: <EA6C29C2-1012-4B9C-BE4F-E98E72D6844C@ipeva.fr>

Yes, please,  try another one, juste in case.
For instance, Zoiper works fine.

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

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

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




Le 28/06/2011 ? 12:42, Al Bogner a ?crit :

> local

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From sascha.daniels at amooma.de  Tue Jun 28 15:27:32 2011
From: sascha.daniels at amooma.de (Sascha Daniels)
Date: Tue, 28 Jun 2011 13:27:32 +0200
Subject: [Freeswitch-users] Hooks for own call logs
In-Reply-To: <BANLkTi=RJMmQx7fk1cOXqF_c_RK8MdCo5A@mail.gmail.com>
References: <BANLkTinz0cptgEt3JQ8Eq2FDuNbgsfDdLw@mail.gmail.com>	<BANLkTim3mx89_p2+KmXk0-iLaBAXQC1EGA@mail.gmail.com>	<4DFC96BA.1080306@amooma.de>
	<BANLkTi=RJMmQx7fk1cOXqF_c_RK8MdCo5A@mail.gmail.com>
Message-ID: <4E09BAA4.7040306@amooma.de>

Hi.

Am 24.06.2011 08:52, schrieb Michael Collins:
> Sorry for the late reply...
>
No problem. I am happy to get answers at all.

> Have you tried setting the api_hangup_hook variable?
> http://wiki.freeswitch.org/wiki/Channel_Variables#api_hangup_hook

I have tried that allready, but nothing was called after the originator
has cancelled the call.

Could not see anything in CLI.

I guess I should make the setup more simple and work with a static
dialplan just to test this.

Regards

                Sascha

-- 
AMOOMA GmbH - Bachstr. 124 - 56566 Neuwied  -->  http://www.amooma.de
Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister Montabaur B14998

B?cher: http://das-asterisk-buch.de - http://ruby-auf-schienen.de



From kbdfck at gmail.com  Tue Jun 28 16:01:28 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Tue, 28 Jun 2011 16:01:28 +0400
Subject: [Freeswitch-users] xml_curl directory - reject registration with
	specific error code?
Message-ID: <BANLkTimykzqC_LOuJQ5LDSDt_dGpjOnbKQ@mail.gmail.com>

Hi all!

Is there a way to reject user with specific error code / message when i use
XML directory via curl? For example, I'd like to distinguish between user
blocked / user not found errors and so on. In other words, instead of
returning default directory stub and 404, I want FS to return some specific
code on failed registration attempt.

Thanks!


-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From max.asterisk at gmail.com  Tue Jun 28 16:31:07 2011
From: max.asterisk at gmail.com (Max Alex)
Date: Tue, 28 Jun 2011 18:01:07 +0530
Subject: [Freeswitch-users] Answer issue on inbound call
In-Reply-To: <BANLkTinE1292_p_VZd+D4N6vvEEot8LXRw@mail.gmail.com>
References: <BANLkTin+f=uDMiOouyLWAorPvNTd1Z6xaA@mail.gmail.com>
	<BANLkTim=TOQD1tUvmHjeRuhAR-eKkGYgsw@mail.gmail.com>
	<BANLkTim3pTPV804t_Usk59eG2C56HAyFdA@mail.gmail.com>
	<BANLkTi=Q3qRBCDRARDWp15=Guwyq-jtNLQ@mail.gmail.com>
	<BANLkTinMokGeJU+wVVA0U-oxB97z+c+ZGg@mail.gmail.com>
	<BANLkTinEp1Cu5mB_UHw-Y+mitGsNByR3sQ@mail.gmail.com>
	<BANLkTikcWDbCkTPRow-TMqPN=D3FV+LjWw@mail.gmail.com>
	<BANLkTinE1292_p_VZd+D4N6vvEEot8LXRw@mail.gmail.com>
Message-ID: <BANLkTin5zjPmdaV6Qa9ZM8dYtBGBUmcKOg@mail.gmail.com>

Hi,
Thanks for your reply.
I have enabled logger as per your help.
I have given completed log on following link of pastebin.
http://pastebin.freeswitch.org/16616

You can see this line as it is showing answered and this call is answered on
my cell phone but on softphone it is ringing and i have rejected from there.
2011-06-28 17:47:52.909657 [NOTICE] mod_freetdm.c:1953 Channel [FreeTDM/2:1/
01234567890] has been answered

It is pre answering the cell phone when it is ringing on phone
Waiting for your help.

Thanks,
Max Alex
Voip Developer



On Mon, Jun 27, 2011 at 9:15 PM, Michael Collins <msc at freeswitch.org> wrote:

> Get a complete, unedited, unabridged console debug log w/ siptrace and put
> it on pastebin w/ "FreeSWITCH Log" for the syntax highlighting. Use "sofia
> global siptrace on" to make sure you can all SIP traffic.
>
> -MC
>
>
> On Mon, Jun 27, 2011 at 6:05 AM, Max Alex <max.asterisk at gmail.com> wrote:
>
>> Hi,
>> Thanks for reply,
>> I have tried the same way and reloaded freeswitch, but still it is
>> answered on first ring of the call.
>> When it is ringing the call on 1001 and the same time it is answered on
>> cell phone, so something is done when it is ringing on 1001.
>>
>> Here is the dialplan for the same
>>
>>     <extension name="Local_Extension">
>>       <condition field="destination_number" expression="^(10[01][0-9])$">
>>         <action application="set" data="dialed_extension=$1"/>
>>         <action application="export" data="dialed_extension=$1"/>
>>         <action application="bind_meta_app" data="2 b s
>> record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>>         <action application="bind_meta_app" data="3 b s
>> execute_extension::cf XML features"/>-->
>>         <action application="set" data="ringback=${us-ring}"/>
>>         <action application="set"
>> data="transfer_ringback=$${hold_music}"/>
>>         <action application="set" data="call_timeout=30"/>
>>           <action application="set" data="hangup_after_bridge=true"/>
>>          <action application="set" data="continue_on_fail=true"/>
>>         <action application="hash"
>> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
>>         <action application="hash"
>> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>>         <action application="set"
>> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
>> var callgroup)}"/>
>>         <action application="hash"
>> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>>
>>        <action application="bridge"
>> data="{ignore_early_media=true}user/${dialed_extension}@${domain_name}"/>
>>         <action application="sleep" data="1000"/>
>>         <action application="voicemail" data="default ${domain_name}
>> ${dialed_extension}"/>
>>       </condition>
>>     </extension>
>>
>>
>> Please help me for this issue.
>>
>>
>> Thanks,
>> Max Alex
>> Voip Developer
>>
>>
>>
>> On Fri, Jun 24, 2011 at 11:38 AM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Are you using the default dialplan? I think you might just need to ignore
>>> early media on your bridge app. If you are using the default.xml file then
>>> locate "Local_Extension" and the bridge line:
>>>
>>>     <action application="bridge" data="user/${dialed_extension}@
>>> ${domain_name}"/>
>>>
>>> Change it to this, then try again:
>>>
>>>     <action application="bridge"
>>> data="{ignore_early_media=true}user/${dialed_extension}@
>>> ${domain_name}"/>
>>>
>>> If I understand correctly, the "symptom" you are experiencing is the
>>> normal operation of the bridge app (and it's cousin, the originate API
>>> command). When the b-leg sends back media, including ringing, the bridge (or
>>> the originate) will be considered "successful," and in the case of bridge,
>>> the b-leg's audio (the early media) will be connected to the a-leg. If you
>>> set ignore_early_media=true then you are explicitly telling the bridge app
>>> that you only want to connect the b-leg to the a-leg if the b-leg actually
>>> answers.
>>>
>>> I hope that made sense...
>>>
>>> -MC
>>>
>>>
>>>
>>> On Thu, Jun 23, 2011 at 9:32 PM, Max Alex <max.asterisk at gmail.com>wrote:
>>>
>>>> Hi,
>>>> Thanks for reply.
>>>> Current scenario is below.
>>>>
>>>> PSTN call -> sangoma device -> freeswitch incoming context -> routed to
>>>> 1001 -> ringing (Answered on cellphone)
>>>> Here when it is routed to 1001 the call it is started ringing, but on
>>>> phone that call is answered and hearding sound of ringing.
>>>>
>>>> Required flow:
>>>> PSTN call -> sangoma device -> freeswitch incoming context -> routed to
>>>> 1001 -> ringing (Ringing on cellphone)
>>>>
>>>> I hope it is understable, the call should not be answered until 1001
>>>> answer it, right not when it is started ring it is answered on cell phone.
>>>> that should not be happen as it is not answered yet.
>>>>
>>>> Waiting for your reply.
>>>>
>>>>
>>>> Thanks,
>>>> Max Alex
>>>> Voip Developer
>>>>
>>>>
>>>>
>>>> On Fri, Jun 24, 2011 at 5:48 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>> I'm not sure I understand the problem. What is happening vs. what you
>>>>> believe should be happening?
>>>>> -MC
>>>>>
>>>>>
>>>>> On Thu, Jun 23, 2011 at 3:31 AM, Max Alex <max.asterisk at gmail.com>wrote:
>>>>>
>>>>>> Hi,
>>>>>> Thanks for your reply.
>>>>>> Here is my configuration and log
>>>>>> http://pastebin.freeswitch.org/16571
>>>>>>
>>>>>> I am using A200 analog sangoma device with freeswitch, it is working
>>>>>> fine but when it is routing call to 1001 then it is answered.
>>>>>> Please provider your suggestions.
>>>>>>
>>>>>> Thanks,
>>>>>> Max Alex
>>>>>> Voip Developer
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Wed, Jun 22, 2011 at 8:51 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>>
>>>>>>> I thought the A200 was an analog card? Maybe I have my numbers mixed
>>>>>>> up...
>>>>>>>
>>>>>>> Go ahead and collect a debug log of this call. It might help to have
>>>>>>> your configs posted as well. Use pastebin.freeswitch.org. See this
>>>>>>> wiki article for tips on how to collect information:
>>>>>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>>>>>>
>>>>>>> -MC
>>>>>>>
>>>>>>> On Wed, Jun 22, 2011 at 3:23 AM, Max Alex <max.asterisk at gmail.com>wrote:
>>>>>>>
>>>>>>>> Hi,
>>>>>>>> I have installed freeswitch latest version with sangoma card A200 as
>>>>>>>> well,
>>>>>>>> I have installed and configured freetdm module with wanpipe drivers
>>>>>>>> for freeswitch,
>>>>>>>> We are properly receiving the inbound calls in public context and
>>>>>>>> then we are routing that call to 1001 extension,
>>>>>>>> it is properly routing to 1001 as well, but we have one issue while
>>>>>>>> routing on 1001.
>>>>>>>>
>>>>>>>> Here is the issue description.
>>>>>>>> I am calling from my cell phone to that DID number of pri line, and
>>>>>>>> then it will start ringing on 1001 extension,
>>>>>>>> When 1001 extension start ringing the call is answered on my cell
>>>>>>>> phone,
>>>>>>>> it is something like freeswitch preanswer or autoanswer the call,
>>>>>>>> how can i stop this answer call when it is ringing on 1001 extension,
>>>>>>>> Waiting for good reply.
>>>>>>>>
>>>>>>>> Thanks,
>>>>>>>> Max Alex
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
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>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
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>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
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>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
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>
> _______________________________________________
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> http://www.cluecon.com 877-7-4ACLUE
>
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From anthony.minessale at gmail.com  Tue Jun 28 17:54:49 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 28 Jun 2011 08:54:49 -0500
Subject: [Freeswitch-users] Phrase macros in conferences
In-Reply-To: <B8CF9984153543B69B76337FDDD3BA3F@stor1>
References: <FE02E1C7-64EA-4E85-AC33-B491AB2FF555@bryansmart.com>
	<BANLkTikRKyF8eppAtqd6DbQqRH29JLnQ7w@mail.gmail.com>
	<93AE9C49-C6F6-44E6-BC7E-5A31AC659C4C@bryansmart.com>
	<9CE9932A-AD27-4183-9C2E-75148116A23B@tech21.com>
	<22CBC366-3CD3-4429-80B6-961AF1566D1C@bryansmart.com>
	<B8CF9984153543B69B76337FDDD3BA3F@stor1>
Message-ID: <BANLkTimzok9Wb41S10ohmCZ31A9XVMcOsA@mail.gmail.com>

with the new say_string function you can build some stuff, its only
one component of phrase macros but its the best we have.

${say_string en en number pronounced 60}

This evals to:

file_string://digits/60.wav

As I was writing this reply I added something to latest get that may
make it a little easier to mix with other files.

${say_string en en number pronounced ~60}

This evals to:

digits/60.wav

so you could chain it together with your own file_string url

file_string://youhave.wav!${say_string en en number pronounced ~60}!messages.wav

This evals to:

file_string://youhave.wav!digits/60.wav!messages.wav


So you can eval and build these in the conference.





On Tue, Jun 28, 2011 at 4:49 AM, Kris <kris at livecall.com> wrote:
> The conference controls/profiles cant be changed after the conference
> starts. Do You know C? We all have to do some C when Freeswitch can't do
> what we want.A feature would have to be added to mod_conference to play
> files unpon joining the conference. It would have to be something like this:
> <action application="conference"
> data="${forum_ext}-${domain_name}@${conference_controls},PlayOnEntry=${name_var}&conf-enter.wav,PlayOnExit=???"/>
>
> To really do the fancier stuff, you have to capture the CUSTOM event. Just
> do it with 2 API calls. Both are scheduled to play one after the other.
> Api.Execute("conference", ConferenceName + " play " + "name.wav");
> //+MemberID);
>
> Api.Execute("conference", ConferenceName + " play " + "conf-enter.wav");
> //+MemberID);
>
> ----- Original Message -----
> From: "Bryan Smart" <bryansmart at bryansmart.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, June 27, 2011 1:03 PM
> Subject: Re: [Freeswitch-users] Phrase macros in conferences
>
>
> Yes, but not sure how I'd do that. In the conference profile, I can play a
> single sound file for an enter/leave. I can't join two files with "!" to
> have them play in series, like "name.wav!conference/conf-enter.wav".
>
> Bryan
>
> On Jun 27, 2011, at 3:37 PM, Brad Mina wrote:
>
> You could have users prompted to record their name to get around the first
> issue and play it back before "... Has joined the conference" is played
> back.
>
> Sent from my iPhone
>
> On Jun 27, 2011, at 12:05 PM, Bryan Smart
> <bryansmart at bryansmart.com<mailto:bryansmart at bryansmart.com>> wrote:
>
> Thanks, Michael. At least I know that it is a limitation, and not me
> overlooking something. :) I have 3 or 4 years on Asterisk, but only months
> on Freeswitch, so I'm always concerned that I haven't dug enough before
> posting requests.
>
> For phrases inside conferences, I imagine 3 cases.
>
> First are dynamic announcements, such as those that announce when someone
> joins or leaves a conference. It is helpful if the joining or leaving party
> can be identified. I imagined combining TTS for the name, along with a
> pre-recorded fragment like conference/conf-has_joined.wav. The result would
> be like "John has entered the conference." Right now, I must use entirely
> TTS for such announcements. I can't use "!" to concatenate files, and I
> can't defer to a phrase macro to string TTS and recordings together.
>
> Second applies to localization. In a dynamic announcement, it might not be
> appropriate for the noun to be spoken first in languages other than English.
> Right now, the conference's prompt language must be customized by creating a
> different conference profile that specifies the correct sound base for the
> language. That could mean that the number of profiles in use has to be
> multiplied by the number of languages that should be supported. I know that
> the profile can omit the sound base, and the sound base of the first person
> to join will be used, but, particularly in a situation that concatenates TTS
> and audio, that can be worse. If the first person to join automatically
> selects sound files for a language, but the concatenation order is different
> for that language, then any rules in that profile that concatenate TTS and
> recordings to render announcements could result in strange outcomes. As far
> as I've been able to discover, the phrase system is the appropriate
> Freeswitch way to localize announcements, so it would probably be best for
> Conference to work that way also, rather than depending on per-case
> localization being handled with multiple profiles. If phrases were
> supported, then a single conference profile could play both static and
> dynamic prompts, and could automatically adjust to any language where phrase
> macros are present on the system.
>
> Last are cases when multiple types of notification are desired. For example,
> notifications might use a combination of cues and speech. An audio cue
> serves a similar function to a visual icon: attracts attention and conveys
> general meaning. In a situation like a conference, there might be soft audio
> cues for joining, leaving, when a new person has the floor, etc. When one of
> those events takes place, the cue plays first to attract attention and set
> the listener's expectations regarding the type of spoken information that
> will follow. Next, an announcement is heard with details about the event.
> "<cue> John has left ?the conference."
>
>
>
>
>
>
> On Jun 27, 2011, at 11:25 AM, Michael Collins wrote:
>
> Phrase macros are not supported inside mod_conference. If you look inside
> mod_conference.c you will see that it has its own functions for injecting
> audio into the conference and it does not support phrase macros. You can
> open up a feature request on Jira. I honestly don't know how easy/difficult
> this would be, so you may consider offering a bounty to sweeten the deal.
>
> Curious: what types of phrases are you trying to play into the conference?
>
> -MC
>
> On Sun, Jun 26, 2011 at 4:57 PM, Bryan Smart
> <<mailto:bryansmart at bryansmart.com>bryansmart at bryansmart.com<mailto:bryansmart at bryansmart.com>>
> wrote:
> I'm attempting to use phrase macros in mod_conference, but I'm not having
> success. Perhaps they aren't supported?
>
> I have a working phrase macro. In the dialplan, both
>
> <action application="phrase" data="my_phrase"/>
>
> and
>
> <action application="playback" data="phrase:my_phrase"/>
>
> Work just fine.
>
> The wiki says that, in conference.conf.xml, entries like pin-sound accept,
> for their value, file paths, strings prefixed with "say" for TTS, and
> strings prefixed with "tone_stream". This looks almost the same as what can
> be passed to the playback application in the dialplan. When I try to prefix
> a string with "phrase", it doesn't work.
>
> <param name="pin-sound" value="phrase:my_phrase"/>
>
> The console tells me "unknown file format". The console still tells me this
> if I make up phrase names
>
> <param name="pin-sound" value="phrase:not_a_real_phrase"/>
>
> So it is complaining about the syntax or capability, rather than an
> unrecognized format of an actual file.
>
> If mod_conference doesn't support phrase macros, is this a near-future
> feature? Phrases would seem to be necessary to ocalize announcements and
> play announcements that are stitched together from multiple files.
>
> Bryan
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> <http://www.cluecon.com/>http://www.cluecon.com<http://www.cluecon.com/>
> 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> <mailto:FreeSWITCH-users at lists.freeswitch.org>FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
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> <http://www.freeswitch.org/>http://www.freeswitch.org<http://www.freeswitch.org/>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> <http://www.cluecon.com/>http://www.cluecon.com<http://www.cluecon.com/>
> 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
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>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com<http://www.cluecon.com/> 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org<http://www.freeswitch.org/>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From avi at avimarcus.net  Tue Jun 28 18:10:09 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 28 Jun 2011 17:10:09 +0300
Subject: [Freeswitch-users] ESL Send MESSAGE_WAITING Error - Cannot find
	profile
Message-ID: <BANLkTimGcbOiDUDgkRxbHZyPYASTidbVjQ@mail.gmail.com>

I'm trying to send a MESSAGE_WAITING via ESL on multi-tenant, but I'm
getting this error:

2011-06-27 10:46:25.443081 [ERR] sofia_presence.c:405 Cannot find profile
20064.domain.com]

the cli lists the domain as an alias to the internal profile.

what should I do?

I am writing some test code in order to try to send MWI events using the ESL
Manager code DLL. Here is my test code:



static void InboundMode2(Object stateInfo)

    {

        //Initializes a new instance of ESLconnection, and connects to the
host $host on the port $port, and supplies $password to freeswitch

        ESLconnection eslConnection = new ESLconnection(myinfo);



        if (eslConnection.Connected() != ESL_SUCCESS)

        {

            Console.WriteLine("Error connecting to FreeSwitch");

            return;

        }



        //Set log level

        //ESL.eslSetLogLevel((int)enLogLevel.DEBUG);



        eslConnection.Api("reloadxml", string.Empty);



        // Subscribe to all events

        ESLevent eslEvent2 = eslConnection.SendRecv("event plain ALL");



        if (eslEvent2 == null)

        {

            Console.WriteLine("Error subscribing to all events");

            return;

        }

        ESLevent eslEvent = new ESLevent("MESSAGE_WAITING", null);

        eslEvent.AddHeader("MWI-Messages-Waiting", "yes");

        eslEvent.AddHeader("MWI-Message-Account",
"103 at 20064.domain.com<103 at 20064.cmvtesttele.com>
");

        eslEvent.AddHeader("MWI-Voice-Message", "1/1 (1/1)");



        eslEvent = eslConnection.SendEvent(eslEvent);

        if (eslEvent == null)

        {

            Console.WriteLine("event error");

            return;

        }



        //Turns an event into colon-separated 'name: value' pairs. The
format parameter isn't used

        Console.WriteLine(eslEvent.Serialize(String.Empty));



        // Grab Events until process is killed

        while (eslConnection.Connected() == ESL_SUCCESS)

        {

            eslEvent = eslConnection.RecvEvent();

            Console.WriteLine(eslEvent.Serialize(String.Empty));

        }

    }



When I send the event, I am getting a debug message from FS as follows:



2011-06-27 10:46:25.443081 [ERR] sofia_presence.c:405 Cannot find profile
20064.domain.com]



Thanks!

-Avi
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From jonas at jonasborjesson.com  Tue Jun 28 11:19:00 2011
From: jonas at jonasborjesson.com (Jonas Borjesson)
Date: Tue, 28 Jun 2011 00:19:00 -0700
Subject: [Freeswitch-users] Copy custom SIP headers -
 {sip_copy_custom_headers=true} doesn't seem to be working
Message-ID: <BANLkTikNA_oA_yswO+7KZ5ir7YUVDNjM2A@mail.gmail.com>

Hi all,

As many others, I need to copy customer X-headers from leg B back over
to leg A and have found a few threads talking about using the
sip_copy_custom_headers=true with the dial string but this does not
seem to be working. I am doing exactly what is described in the thread
below but like that guy, it does not work for me either and that
thread never came to any conclusion. Does anyone know the status of
this feature or what I may be doing wrong? Any help is greatly
appreciated!

Of course, I tried on the latest and greatest FS -
1.0.head-git-4962542 2011-06-27 10-15-03 -0500

The tread talking about this very feature:
http://comments.gmane.org/gmane.comp.telephony.freeswitch.user/36287

And documentation:
http://wiki.freeswitch.org/wiki/Variable_sip_copy_custom_headers

Thanks,

/Jonas


From mosbah.abdelkader at gmail.com  Tue Jun 28 18:28:38 2011
From: mosbah.abdelkader at gmail.com (mosbah abdelkader)
Date: Tue, 28 Jun 2011 15:28:38 +0100
Subject: [Freeswitch-users] Forwarding SIP calls between 2 VoIP carriers
	without using SIP TRUNKING
Message-ID: <BANLkTikAU0ouo+9qWHyaHdMPkDdCi=uYaw@mail.gmail.com>

Hi,

One client have 2 SIP accounts: SIP1 and SIP2 with 2 different carriers: C1
and C2.

He wants to forward all calls destined to SIP1 to SIP2. C2 is using
FreeSWITCH.

I don't want to register SIP1 as SIP trunk in C2 because if the number of
SIP accounts is big, there will be a performance problem as each account
will be registered as a SIP trunk.

Can anyone help me by giving an idea and a config sample for FreeSWITCH.

Thank you.

--
Best Regards
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From msc at freeswitch.org  Tue Jun 28 18:59:39 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 28 Jun 2011 07:59:39 -0700
Subject: [Freeswitch-users] UniMRCP Server as daemon won't connect RTSP
In-Reply-To: <361E98F99D3CC3439EED59BC1924ED6950815F@SERVER2003.SecuReachSystems.local>
References: <361E98F99D3CC3439EED59BC1924ED6950815F@SERVER2003.SecuReachSystems.local>
Message-ID: <BANLkTimMYtmSvWvVbM3tODRBgRocMSxcwA@mail.gmail.com>

How about running unimrcpserver inside of GNU screen? It may not be a
perfect solution but it might let you keep working until you figure out why
the daemon isn't working.
-MC

On Mon, Jun 27, 2011 at 2:52 PM, Jason Moran <jmoran at secureachsystems.com>wrote:

>  I?ve been playing around with IVONA?s new unimrcp plugin with unimrcp
> 1.0.0 (r1725) on OpenSuse 11.4****
>
> ** **
>
> Anyway, when I start it up with the following, it stays in the foreground
> of my SSH w/ the following command:****
>
> ./unimrcpserver****
>
> That works! However, since it?s in the foreground when I close the SSH
> window, it kills the process.****
>
> ** **
>
> If I use the so-called daemon mode (-d or --daemon) it says it is going
> into daemon mode, but FS will immediately return that it ?Failed to Connect
> to RTSP Server...? at the IP:port I specified. The unimrcpserver process is
> running, but doesn?t seem to respond to anything.****
>
> ./unimrcpserver ?d****
>
> ./unimrcpserver --daemon****
>
> ** **
>
> If I attempt to background it by either using ?&? or ctrl-z it says [1]+
> Stopped  ./unimrcpserver and FS will successfully make the RTSP connection
> but then nothing will happen. Mod_unimrcp will spit out a warning about
> ?MRCP session has not opened after 5000 ms?****
>
> ./unimrcpserver &****
>
> Those are hard to kill. But when I kill -9 it then FS will finally remove
> the MRCP handle and tell me that it couldn?t allocate the speech engine.**
> **
>
> ** **
>
> Lastly, I can nohup it, but then unimrcpserver eats up 95%+ of the CPU
> (instead of 1-3% as it does when I have it in the foreground), which it does
> not do when it runs in any other modes. It also makes a very, very large
> nohup.out file that keeps on growing. Even if I tell it to not log.****
>
> nohup ./unimrcpserver &****
>
> nohup ./unimrcpserver ?o 0 &****
>
> ** **
>
> Ideas??
> Thanks,****
>
> Jason Moran****
>
> ** **
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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>
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From jmoran at secureachsystems.com  Tue Jun 28 19:18:33 2011
From: jmoran at secureachsystems.com (Jason Moran)
Date: Tue, 28 Jun 2011 11:18:33 -0400
Subject: [Freeswitch-users] UniMRCP Server as daemon won't connect RTSP
References: <361E98F99D3CC3439EED59BC1924ED6950815F@SERVER2003.SecuReachSystems.local>
	<BANLkTimMYtmSvWvVbM3tODRBgRocMSxcwA@mail.gmail.com>
Message-ID: <361E98F99D3CC3439EED59BC1924ED69508175@SERVER2003.SecuReachSystems.local>

That will technically work, but it is not a good solution. Since SSH is
our only allowed remoting option (other methods have been shut down by
our security guy), I need to find a way for unimrcp daemon mode to be
stable with freeswitch.

 

Can other people successfully connect in daemon mode?

-Jason

 

From: Michael Collins [mailto:msc at freeswitch.org] 
Sent: Tuesday, June 28, 2011 11:00 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] UniMRCP Server as daemon won't connect
RTSP

 

How about running unimrcpserver inside of GNU screen? It may not be a
perfect solution but it might let you keep working until you figure out
why the daemon isn't working.
-MC

On Mon, Jun 27, 2011 at 2:52 PM, Jason Moran
<jmoran at secureachsystems.com> wrote:

I've been playing around with IVONA's new unimrcp plugin with unimrcp
1.0.0 (r1725) on OpenSuse 11.4

 

Anyway, when I start it up with the following, it stays in the
foreground of my SSH w/ the following command:

./unimrcpserver

That works! However, since it's in the foreground when I close the SSH
window, it kills the process.

 

If I use the so-called daemon mode (-d or --daemon) it says it is going
into daemon mode, but FS will immediately return that it "Failed to
Connect to RTSP Server..." at the IP:port I specified. The unimrcpserver
process is running, but doesn't seem to respond to anything.

./unimrcpserver -d

./unimrcpserver --daemon

 

If I attempt to background it by either using "&" or ctrl-z it says [1]+
Stopped  ./unimrcpserver and FS will successfully make the RTSP
connection but then nothing will happen. Mod_unimrcp will spit out a
warning about "MRCP session has not opened after 5000 ms"

./unimrcpserver &

Those are hard to kill. But when I kill -9 it then FS will finally
remove the MRCP handle and tell me that it couldn't allocate the speech
engine.

 

Lastly, I can nohup it, but then unimrcpserver eats up 95%+ of the CPU
(instead of 1-3% as it does when I have it in the foreground), which it
does not do when it runs in any other modes. It also makes a very, very
large nohup.out file that keeps on growing. Even if I tell it to not
log.

nohup ./unimrcpserver &

nohup ./unimrcpserver -o 0 &

 

Ideas??
Thanks,

Jason Moran

 


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

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From msc at freeswitch.org  Tue Jun 28 19:18:12 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 28 Jun 2011 08:18:12 -0700
Subject: [Freeswitch-users] Proper prompt gain/level
In-Reply-To: <1BB0CF88-DA61-4AB6-9CC6-328CCB9846AD@bryansmart.com>
References: <FE081E27-8D62-4715-B7AA-64432598E013@bryansmart.com>
	<0D3AEFF37C1643B79CE3F7A7EE9EAEC3@dell9400>
	<66E81300-46E9-4C81-B2E0-123BCADDCAEB@bryansmart.com>
	<4E088A24.7010002@coppice.org>
	<3081AB39-6EF0-4C43-9AC1-7D43FCD378CD@bryansmart.com>
	<BANLkTikTK3JW=jDFNqkZqPxhic_bzL=uCQ@mail.gmail.com>
	<BANLkTimSuFpBXwANu8qc2q24z-xNYdcJnw@mail.gmail.com>
	<1BB0CF88-DA61-4AB6-9CC6-328CCB9846AD@bryansmart.com>
Message-ID: <BANLkTi=9-B3R74voiHLV6RgcvvkqwS6aZQ@mail.gmail.com>

You can download the 48kHz files from files.freeswitch.org. The script
itself is nothing special - all it does is cycle through the target sampling
rates and run each file through sox. Here's a copy of the script:

base_dir="48000"
rates="48000 32000 16000 8000"
version="1.0.16"
voice="en-us-callie"
voice_dir="en/us/callie"
tar_path="../../.."
tmp_dir="tmp"
CWD=`pwd`

for rate in $rates; do \
  mkdir -p $tmp_dir/$voice_dir
  for dir in `ls $base_dir`; do \
    test -d $tmp_dir/$voice_dir/$dir/$rate || mkdir -p
$tmp_dir/$voice_dir/$dir/$rate; \
    for filename in `ls $base_dir/$dir`; do \
      echo sox -v 0.2 $base_dir/$dir/$filename -r $rate -c 1
$tmp_dir/$voice_dir/$dir/$rate/$filename; \
      sox -v 0.2 $base_dir/$dir/$filename -r $rate -c 1
$tmp_dir/$voice_dir/$dir/$rate/$filename; \
    done ; \
  done ; \
  cd $tmp_dir
  echo tar -cvzf
$CWD/$tar_path/freeswitch-sounds-$voice-$rate-$version.tar.gz *; \
  tar -cvzf $CWD/$tar_path/freeswitch-sounds-$voice-$rate-$version.tar.gz *;
\
  cd -
  rm -rf $tmp_dir
done

cd $tar_path
for rate in $rates; do \
  openssl dgst -sha1 freeswitch-sounds-$voice-$rate-$version.tar.gz >
freeswitch-sounds-$voice-$rate-$version.tar.gz.sha1 ; \
  openssl dgst -md5 freeswitch-sounds-$voice-$rate-$version.tar.gz >
freeswitch-sounds-$voice-$rate-$version.tar.gz.md5 ; \
done
cd $CWD

Let me know if you have any suggestions.
-MC

On Mon, Jun 27, 2011 at 8:25 PM, Bryan Smart <bryansmart at bryansmart.com>wrote:

> -v is a multiplier. So, -v 0.2 reduces gain to 20% of original.
>
> It is typical to supply professionally recorded audio with peaks boosted to
> 0DBFS. That is probably how the prompts were originally supplied. It makes
> since that -v 0.2 would reduce the max power to 20%, around -15 to -16 DBFS.
>
> The FS build process downloads pre-processed versions of these files. Is
> there somewhere online that I can find the source recordings+script that
> produces the pre-processed files that are downloaded by the build process?
>
> Bryan
>
> On Jun 27, 2011, at 7:32 PM, Michael Collins wrote:
>
>
>
> On Mon, Jun 27, 2011 at 3:07 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Have a look at the 48khz versions of the files, those should be the
>> originals.
>> Someone runs a batch sox command on them to get them to the other levels.
>>
>
> Just for the record, the sox command used to normalize the sounds is:
> sox -v 0.2 $file_in -r $rate -c 1 $file_out
>
> The -v format is the "volume" format and if I read the sox man page
> correctly it means a "linear amplitude adjustment". What I can't tell from
> the man page is whether .2 means "reduce by 20%" or "reduce to 20%" or
> something else. In any case, like Tony says, we are not picky about this as
> long as it's not pointlessly loud.
>
> -MC
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mays.david at gmail.com  Tue Jun 28 19:38:37 2011
From: mays.david at gmail.com (David Ma)
Date: Tue, 28 Jun 2011 23:38:37 +0800
Subject: [Freeswitch-users] Leg-A is automatically disconnected on Leg-B
 orginate failure
In-Reply-To: <BANLkTimiExLew_1cjMuQVpF3CKkR8mrGXQ@mail.gmail.com>
References: <1308213151543-6482192.post@n2.nabble.com>
	<BANLkTi=xYgBTPVGxVGmk9f2=bmX5X2vF=Q@mail.gmail.com>
	<BANLkTikhPmjN5HeeJHBnQt1ivA8zbtCnYA@mail.gmail.com>
	<BANLkTi=m7oMix1ZRC9URtRuAbsG2fWyhMQ@mail.gmail.com>
	<BANLkTimxBU8m311LVKMWcFOjukwLg+ZUGQ@mail.gmail.com>
	<BANLkTinZJf3BxB7QHefnmOvxQmVW2r4dTg@mail.gmail.com>
	<BANLkTimiExLew_1cjMuQVpF3CKkR8mrGXQ@mail.gmail.com>
Message-ID: <BANLkTi=FJM-7x=eeFodBB7MK+s1bWRO-Xw@mail.gmail.com>

Hi Michael,

Sorry for the inconvenience. I just pastebin the log. It can be found in
path http://pastebin.freeswitch.org/16617.

Thanks,
David Ma

On Mon, Jun 27, 2011 at 11:09 PM, Michael Collins <msc at freeswitch.org>wrote:

> This makes my eyes bleed. Can you please put this on
> pastebin.freeswitch.org? Use "FreeSWITCH Log" as the syntax highlight.
> -MC
>
> On Sun, Jun 26, 2011 at 11:27 PM, David Ma <mays.david at gmail.com> wrote:
>
>> Hi Michael,
>>
>> Thanks very much for the response. There is no dialplan specified for this
>> call. In the FS setting, G729 is used as preferred stack for originating
>> calls. The leg-A and B are bridged immediately after receiving
>> PROGRESS-MEDIA from leg-B.
>>
>> Calls are originated with following parameters:
>> *
>> -- Leg-A --
>>             "api originate
>> {origination_caller_id_number=<caller>,sip_cid_type=pid,privacy=yes,continue_on_fail=true}
>> <called> &park()"
>>
>> **-- Leg-B --*
>> *            "api originate
>> {origination_caller_id_number=<caller>,originate_timeout=60,sip_cid_type=pid,privacy=yes,continue_on_fail=false}
>> <called> &park()"
>> *
>>
>> The entire debug log for this call follows.
>>
>> Thanks,
>> D.Ma
>>
>> On Fri, Jun 24, 2011 at 2:41 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Pastebin the entire debug log, including the siptrace. Also include the
>>> originate line and any other dialplan config that might be used.
>>> -MC
>>>
>>>
>>> On Thu, Jun 23, 2011 at 11:26 PM, David Ma <mays.david at gmail.com> wrote:
>>>
>>>> Hi Michael,
>>>>
>>>> Unfortunately this problem still happens.
>>>>
>>>> I enabled "continue_on_fail" for leg-A when I originated the call. Leg-A
>>>> call went well. Then I originated leg-B call ("continue_on_fail" is NOT set
>>>> for leg-B), which failed for [DESTINATION_OUT_OF_ORDER]. As the consequence,
>>>> leg-A was hung up by FS automatically for [ORIGINATOR_CANCEL].
>>>>
>>>> The log excerpt follows.
>>>>
>>>> Do you think "continue_on_fail" should be also enabled for leg-B call?
>>>>
>>>> Thanks,
>>>> D.Ma
>>>>
>>>> 2011-06-24 13:29:35.887830 [DEBUG] switch_ivr_originate.c:1971 variable
>>>> string 5 = [continue_on_fail=true]
>>>>
>>>> 2011-06-24 13:29:35.887830 [NOTICE] switch_channel.c:808 New Channel
>>>> sofia/external/03996597632298 at 203.208.207.212[c0bd700d-913c-42ad-b68f-81001bf658b8]
>>>> [...]
>>>> 2011-06-24 13:29:48.142264 [DEBUG] switch_core_state_machine.c:372
>>>> (sofia/external/03996563750911 at 203.208.207.212) State SOFT_EXECUTE
>>>> going to sleep
>>>> 2011-06-24 13:29:48.161779 [DEBUG] switch_channel.c:2535
>>>> (sofia/external/03996563750911 at 203.208.207.212) Callstate Change EARLY
>>>> -> HANGUP
>>>> 2011-06-24 13:29:48.161779 [NOTICE] switch_ivr_originate.c:1045 Hangup
>>>> sofia/external/03996563750911 at 203.208.207.212 [CS_SOFT_EXECUTE]
>>>> [DESTINATION_OUT_OF_ORDER]
>>>> 2011-06-24 13:29:48.161779 [DEBUG] switch_channel.c:2551 Send signal
>>>> sofia/external/03996563750911 at 203.208.207.212 [KILL]
>>>> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_session.c:1116 Send
>>>> signal sofia/external/03996563750911 at 203.208.207.212 [BREAK]
>>>> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:320
>>>> (sofia/external/03996563750911 at 203.208.207.212) Running State Change
>>>> CS_HANGUP
>>>> 2011-06-24 13:29:48.161779 [DEBUG] switch_channel.c:2535
>>>> (sofia/external/03996597632298 at 203.208.207.212) Callstate Change ACTIVE
>>>> -> HANGUP
>>>> 2011-06-24 13:29:48.161779 [NOTICE] switch_ivr_bridge.c:772 Hangup
>>>> sofia/external/03996597632298 at 203.208.207.212 [CS_SOFT_EXECUTE]
>>>> [ORIGINATOR_CANCEL]
>>>> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_state_machine.c:557
>>>> (sofia/external/03996563750911 at 203.208.207.212) State HANGUP
>>>> 2011-06-24 13:29:48.161779 [DEBUG] mod_sofia.c:457 Channel
>>>> sofia/external/03996563750911 at 203.208.207.212 hanging up, cause:
>>>> DESTINATION_OUT_OF_ORDER
>>>> 2011-06-24 13:29:48.161779 [DEBUG] mod_sofia.c:510 Sending CANCEL to
>>>> sofia/external/03996563750911 at 203.208.207.212
>>>> 2011-06-24 13:29:48.161779 [DEBUG] switch_channel.c:2551 Send signal
>>>> sofia/external/03996597632298 at 203.208.207.212 [KILL]
>>>> 2011-06-24 13:29:48.161779 [DEBUG] switch_core_session.c:1116 Send
>>>> signal sofia/external/03996597632298 at 203.208.207.212 [BREAK]
>>>>
>>>> On Fri, Jun 17, 2011 at 10:52 AM, David Ma <mays.david at gmail.com>wrote:
>>>>
>>>>> Hi Michael,
>>>>>
>>>>> Thanks very much for your prompt response! I appreciate the information
>>>>> provided.
>>>>>
>>>>> I was actually searching the the existence of such a variable. I was
>>>>> not so luck to find it out and thereby resort to the support forum.
>>>>>
>>>>> I've modified my code to build this parameter into my application. Will
>>>>> feedback to you after verification.
>>>>>
>>>>> Thanks again,
>>>>> D.Ma
>>>>>
>>>>> On Fri, Jun 17, 2011 at 4:51 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>
>>>>>> How about setting this?
>>>>>> http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
>>>>>>
>>>>>> -MC
>>>>>>
>>>>>>
>>>>>> On Thu, Jun 16, 2011 at 1:32 AM, dma <mays.david at gmail.com> wrote:
>>>>>>
>>>>>>> I am creating a call-back solution. After leg-A answers, I originate
>>>>>>> leg-B
>>>>>>> call. After receiving SIP 183 from Leg-B, I bridge the 2 legs.
>>>>>>> However, in
>>>>>>> some cases, leg-A is automatically disconnected by FreeSwitch on
>>>>>>> leg-B
>>>>>>> failure, for example, DESTINATION_OUT_OF_ORDER. The application is
>>>>>>> not given
>>>>>>> a chance to handle leg-B failure event. This should not be a correct
>>>>>>> scenario because I never set "hangup-after-bridge", which is false by
>>>>>>> default.
>>>>>>>
>>>>>>> The right way should be, FreeSwitch doesn't hang up leg-A
>>>>>>> automatically, but
>>>>>>> give a chance for the application to decide what to do.
>>>>>>>
>>>>>>> Please see the logs below:
>>>>>>>
>>>>>>> =================================================
>>>>>>>
>>>>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971
>>>>>>> variable
>>>>>>> string 0 = [origination_caller_id_number=03996563750914]
>>>>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971
>>>>>>> variable
>>>>>>> string 1 = [originate_timeout=30]
>>>>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971
>>>>>>> variable
>>>>>>> string 2 = [ccd_session_id=20110610105829676824]
>>>>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971
>>>>>>> variable
>>>>>>> string 3 = [sip_cid_type=pid]
>>>>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_ivr_originate.c:1971
>>>>>>> variable
>>>>>>> string 4 = [privacy=yes]
>>>>>>> 2011-06-10 11:09:55.370506 [NOTICE] switch_channel.c:808 New Channel
>>>>>>> sofia/external/03996590031055 at 203.208.207.212
>>>>>>> [ea57b74b-a8c2-4fea-9683-98054dc03a79]
>>>>>>> 2011-06-10 11:09:55.370506 [DEBUG] mod_sofia.c:4129
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_NEW
>>>>>>> ->
>>>>>>> CS_INIT
>>>>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_core_session.c:1116 Send
>>>>>>> signal
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_core_state_machine.c:320
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>>>> CS_INIT
>>>>>>> 2011-06-10 11:09:55.370506 [DEBUG] switch_core_state_machine.c:356
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State INIT
>>>>>>> 2011-06-10 11:09:55.370506 [DEBUG] mod_sofia.c:84
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA INIT
>>>>>>> send 984 bytes to udp/[203.208.207.212]:5060 at 03:09:55.477997:
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>>   INVITE sip:03996590031055 at 203.208.207.212 SIP/2.0
>>>>>>>   Via: SIP/2.0/UDP 202.73.56.46:5080
>>>>>>> ;rport;branch=z9hG4bKXjQ7eFpKypy5D
>>>>>>>   Max-Forwards: 70
>>>>>>>   From: "" &lt;sip:03996563750914 at 202.73.56.46&gt;;tag=H4ZN43UH9S9Qc
>>>>>>>   To: &lt;sip:03996590031055 at 203.208.207.212&gt;
>>>>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>>>>   CSeq: 13501633 INVITE
>>>>>>>   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
>>>>>>>   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
>>>>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>>>>> REGISTER, REFER, NOTIFY
>>>>>>>   Supported: timer, precondition, path, replaces
>>>>>>>   Allow-Events: talk, hold, refer
>>>>>>>   Privacy: none
>>>>>>>   Content-Type: application/sdp
>>>>>>>   Content-Disposition: session
>>>>>>>   Content-Length: 204
>>>>>>>   X-FS-Support: update_display
>>>>>>>   P-Asserted-Identity: "" &lt;sip:03996563750914 at 202.73.56.46&gt;
>>>>>>>
>>>>>>>   v=0
>>>>>>>   o=FreeSWITCH 1307645395 1307645396 IN IP4 202.73.56.46
>>>>>>>   s=FreeSWITCH
>>>>>>>   c=IN IP4 202.73.56.46
>>>>>>>   t=0 0
>>>>>>>   m=audio 30000 RTP/AVP 18 3 101 13
>>>>>>>   a=rtpmap:101 telephone-event/8000
>>>>>>>   a=fmtp:101 0-16
>>>>>>>   a=ptime:20
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>> 2011-06-10 11:09:55.371674 [DEBUG] mod_sofia.c:124
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change CS_INIT
>>>>>>> ->
>>>>>>> CS_ROUTING
>>>>>>> 2011-06-10 11:09:55.371674 [DEBUG] switch_core_session.c:1116 Send
>>>>>>> signal
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>>>> 2011-06-10 11:09:55.371674 [DEBUG] sofia.c:4646 Channel
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 entering state
>>>>>>> [calling][0]
>>>>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:356
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State INIT going to
>>>>>>> sleep
>>>>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:320
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>>>> CS_ROUTING
>>>>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_channel.c:1657
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change
>>>>>>> DOWN ->
>>>>>>> RINGING
>>>>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:359
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State ROUTING
>>>>>>> 2011-06-10 11:09:55.373478 [DEBUG] mod_sofia.c:147
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA ROUTING
>>>>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_ivr_originate.c:66
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change
>>>>>>> CS_ROUTING ->
>>>>>>> CS_CONSUME_MEDIA
>>>>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_session.c:1116 Send
>>>>>>> signal
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:359
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State ROUTING going
>>>>>>> to sleep
>>>>>>> 2011-06-10 11:09:55.373478 [DEBUG] switch_core_state_machine.c:320
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>>>> CS_CONSUME_MEDIA
>>>>>>> 2011-06-10 11:09:55.374485 [DEBUG] switch_core_state_machine.c:378
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State CONSUME_MEDIA
>>>>>>> 2011-06-10 11:09:55.374485 [DEBUG] switch_core_state_machine.c:378
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State CONSUME_MEDIA
>>>>>>> going to
>>>>>>> sleep
>>>>>>> recv 307 bytes from udp/[203.208.207.212]:5060 at 03:09:55.485130:
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>>   SIP/2.0 100 Trying
>>>>>>>   Via: SIP/2.0/UDP
>>>>>>> 10.1.1.46:5080
>>>>>>> ;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>>>>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>>>>>   To: &lt;sip:03996590031055 at 203.208.207.212&gt;
>>>>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>>>>   CSeq: 13501633 INVITE
>>>>>>>   Content-Length:     0
>>>>>>>
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>> recv 669 bytes from udp/[203.208.207.212]:5060 at 03:09:56.677788:
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>>   SIP/2.0 183 Session Progress
>>>>>>>   Via: SIP/2.0/UDP
>>>>>>> 10.1.1.46:5080
>>>>>>> ;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>>>>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>>>>>   To:
>>>>>>> &lt;sip:03996590031055 at 203.208.207.212
>>>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>>>>   CSeq: 13501633 INVITE
>>>>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER,
>>>>>>> NOTIFY,
>>>>>>> SUBSCRIBE, UPDATE
>>>>>>>   Content-Type: application/sdp
>>>>>>>   Content-Length:   189
>>>>>>>
>>>>>>>   v=0
>>>>>>>   o=- 421265648 1 IN IP4 203.208.207.219
>>>>>>>   s=session
>>>>>>>   c=IN IP4 203.208.207.196
>>>>>>>   t=0 0
>>>>>>>   m=audio 30792 RTP/AVP 18 101
>>>>>>>   a=rtpmap:18 G729/8000
>>>>>>>   a=rtpmap:101 telephone-event/8000
>>>>>>>   a=sendrecv
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>> 2011-06-10 11:09:56.571839 [INFO] sofia.c:729
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 Update Callee ID to
>>>>>>> "Outbound
>>>>>>> Call" <03996590031055>
>>>>>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia.c:4646 Channel
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 entering state
>>>>>>> [proceeding][183]
>>>>>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia.c:4657 Remote SDP:
>>>>>>> v=0
>>>>>>> o=- 421265648 1 IN IP4 203.208.207.219
>>>>>>> s=session
>>>>>>> c=IN IP4 203.208.207.196
>>>>>>> t=0 0
>>>>>>> m=audio 30792 RTP/AVP 18 101
>>>>>>> a=rtpmap:18 G729/8000
>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>
>>>>>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:4467 Audio Codec
>>>>>>> Compare
>>>>>>> [G729:18:8000:20:8000]/[G729:18:8000:20:8000]
>>>>>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:2757 Set Codec
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 G729/8000 20 ms 160
>>>>>>> samples
>>>>>>> 8000 bits
>>>>>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:4565 Set 2833 dtmf
>>>>>>> send
>>>>>>> payload to 101
>>>>>>> 2011-06-10 11:09:56.571839 [DEBUG] sofia_glue.c:2987 AUDIO RTP
>>>>>>> [sofia/external/03996590031055 at 203.208.207.212] 10.1.1.46 port 30000
>>>>>>> ->
>>>>>>> 203.208.207.196 port 30792 codec: 18 ms: 20
>>>>>>> 2011-06-10 11:09:56.571839 [DEBUG] switch_rtp.c:1607 Starting timer
>>>>>>> [soft]
>>>>>>> 160 bytes per 20ms
>>>>>>> 2011-06-10 11:09:56.573731 [DEBUG] sofia_glue.c:3228 Set 2833 dtmf
>>>>>>> send
>>>>>>> payload to 101
>>>>>>> 2011-06-10 11:09:56.573731 [DEBUG] sofia_glue.c:3233 Set 2833 dtmf
>>>>>>> receive
>>>>>>> payload to 101
>>>>>>> 2011-06-10 11:09:56.573731 [NOTICE] sofia_glue.c:3680 Pre-Answer
>>>>>>> sofia/external/03996590031055 at 203.208.207.212!
>>>>>>> 2011-06-10 11:09:56.573731 [DEBUG] switch_channel.c:2627
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change
>>>>>>> RINGING ->
>>>>>>> EARLY
>>>>>>> 2011-06-10 11:09:56.573731 [DEBUG] switch_ivr_originate.c:3408
>>>>>>> Originate
>>>>>>> Resulted in Success: [sofia/external/03996590031055 at 203.208.207.212]
>>>>>>> 2011-06-10 11:09:56.573731 [DEBUG] mod_commands.c:3205
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change
>>>>>>> CS_CONSUME_MEDIA -> CS_EXECUTE
>>>>>>> 2011-06-10 11:09:56.573731 [DEBUG] switch_core_session.c:1116 Send
>>>>>>> signal
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>>>> 2011-06-10 11:09:56.575262 [DEBUG] switch_core_state_machine.c:320
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>>>> CS_EXECUTE
>>>>>>> 2011-06-10 11:09:56.575262 [DEBUG] switch_core_state_machine.c:366
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State EXECUTE
>>>>>>> 2011-06-10 11:09:56.575262 [DEBUG] mod_sofia.c:240
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA EXECUTE
>>>>>>> 2011-06-10 11:09:56.575262 [DEBUG] switch_core_state_machine.c:157
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 Standard EXECUTE
>>>>>>> EXECUTE sofia/external/03996590031055 at 203.208.207.212 park()
>>>>>>> 2011-06-10 11:09:56.618770 [DEBUG] switch_rtp.c:2933 Correct ip/port
>>>>>>> confirmed.
>>>>>>> recv 669 bytes from udp/[203.208.207.212]:5060 at 03:10:01.562021:
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>>   SIP/2.0 183 Session Progress
>>>>>>>   Via: SIP/2.0/UDP
>>>>>>> 10.1.1.46:5080
>>>>>>> ;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>>>>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>>>>>   To:
>>>>>>> &lt;sip:03996590031055 at 203.208.207.212
>>>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>>>>   CSeq: 13501633 INVITE
>>>>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER,
>>>>>>> NOTIFY,
>>>>>>> SUBSCRIBE, UPDATE
>>>>>>>   Content-Type: application/sdp
>>>>>>>   Content-Length:   189
>>>>>>>
>>>>>>>   v=0
>>>>>>>   o=- 421265648 2 IN IP4 203.208.207.219
>>>>>>>   s=session
>>>>>>>   c=IN IP4 203.208.207.196
>>>>>>>   t=0 0
>>>>>>>   m=audio 30792 RTP/AVP 18 101
>>>>>>>   a=rtpmap:18 G729/8000
>>>>>>>   a=rtpmap:101 telephone-event/8000
>>>>>>>   a=sendrecv
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>> 2011-06-10 11:10:01.455944 [INFO] sofia.c:729
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 Update Callee ID to
>>>>>>> "03996590031055" <03996590031055>
>>>>>>> 2011-06-10 11:10:01.455944 [DEBUG] sofia.c:4641 Channel
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 skipping state
>>>>>>> [proceeding][183]
>>>>>>> recv 669 bytes from udp/[203.208.207.212]:5060 at 03:10:01.563178:
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>>   SIP/2.0 183 Session Progress
>>>>>>>   Via: SIP/2.0/UDP
>>>>>>> 10.1.1.46:5080
>>>>>>> ;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>>>>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>>>>>   To:
>>>>>>> &lt;sip:03996590031055 at 203.208.207.212
>>>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>>>>   CSeq: 13501633 INVITE
>>>>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER,
>>>>>>> NOTIFY,
>>>>>>> SUBSCRIBE, UPDATE
>>>>>>>   Content-Type: application/sdp
>>>>>>>   Content-Length:   189
>>>>>>>
>>>>>>>   v=0
>>>>>>>   o=- 421265648 3 IN IP4 203.208.207.219
>>>>>>>   s=session
>>>>>>>   c=IN IP4 203.208.207.196
>>>>>>>   t=0 0
>>>>>>>   m=audio 30792 RTP/AVP 18 101
>>>>>>>   a=rtpmap:18 G729/8000
>>>>>>>   a=rtpmap:101 telephone-event/8000
>>>>>>>   a=sendrecv
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>> 2011-06-10 11:10:01.457169 [DEBUG] sofia.c:4641 Channel
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 skipping state
>>>>>>> [proceeding][183]
>>>>>>> recv 749 bytes from udp/[203.208.207.212]:5060 at 03:10:27.365284:
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>>   SIP/2.0 200 Ok
>>>>>>>   Via: SIP/2.0/UDP
>>>>>>> 10.1.1.46:5080
>>>>>>> ;rport=5080;branch=z9hG4bKXjQ7eFpKypy5D;received=10.1.1.46
>>>>>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>>>>>   To:
>>>>>>> &lt;sip:03996590031055 at 203.208.207.212
>>>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>>>>   CSeq: 13501633 INVITE
>>>>>>>   Contact: <sip:203.208.207.212:5060>
>>>>>>>   Allow-Events: refer
>>>>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER,
>>>>>>> NOTIFY,
>>>>>>> SUBSCRIBE, UPDATE
>>>>>>>   Content-Type: application/sdp
>>>>>>>   Supported: 100rel, timer, replaces
>>>>>>>   Content-Length:   189
>>>>>>>
>>>>>>>   v=0
>>>>>>>   o=- 421265648 4 IN IP4 203.208.207.219
>>>>>>>   s=session
>>>>>>>   c=IN IP4 203.208.207.196
>>>>>>>   t=0 0
>>>>>>>   m=audio 30792 RTP/AVP 18 101
>>>>>>>   a=rtpmap:18 G729/8000
>>>>>>>   a=rtpmap:101 telephone-event/8000
>>>>>>>   a=sendrecv
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>> 2011-06-10 11:10:27.258092 [DEBUG] sofia.c:4646 Channel
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 entering state
>>>>>>> [completing][200]
>>>>>>> 2011-06-10 11:10:27.258092 [DEBUG] sofia.c:4657 Remote SDP:
>>>>>>> v=0
>>>>>>> o=- 421265648 4 IN IP4 203.208.207.219
>>>>>>> s=session
>>>>>>> c=IN IP4 203.208.207.196
>>>>>>> t=0 0
>>>>>>> m=audio 30792 RTP/AVP 18 101
>>>>>>> a=rtpmap:18 G729/8000
>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>
>>>>>>> send 405 bytes to udp/[203.208.207.212]:5060 at 03:10:27.366562:
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>>   ACK sip:203.208.207.212:5060 SIP/2.0
>>>>>>>   Via: SIP/2.0/UDP 202.73.56.46:5080
>>>>>>> ;rport;branch=z9hG4bKyUg0ga7pUZmrS
>>>>>>>   Max-Forwards: 70
>>>>>>>   From: "" &lt;sip:03996563750914 at 202.73.56.46&gt;;tag=H4ZN43UH9S9Qc
>>>>>>>   To:
>>>>>>> &lt;sip:03996590031055 at 203.208.207.212
>>>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>>>>   CSeq: 13501633 ACK
>>>>>>>   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
>>>>>>>   Content-Length: 0
>>>>>>>
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>> 2011-06-10 11:10:27.260267 [DEBUG] sofia.c:4646 Channel
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 entering state
>>>>>>> [ready][200]
>>>>>>> 2011-06-10 11:10:27.260267 [DEBUG] switch_channel.c:2782
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change
>>>>>>> EARLY ->
>>>>>>> ACTIVE
>>>>>>> 2011-06-10 11:10:27.260267 [NOTICE] sofia.c:5175 Channel
>>>>>>> [sofia/external/03996590031055 at 203.208.207.212] has been answered
>>>>>>> 2011-06-10 11:10:27.264178 [DEBUG] switch_scheduler.c:214 Added task
>>>>>>> 23
>>>>>>> switch_ivr_schedule_hangup (ea57b74b-a8c2-4fea-9683-98054dc03a79) to
>>>>>>> run at
>>>>>>> 1307676927
>>>>>>> 2011-06-10 11:10:27.265202 [DEBUG] switch_core_session.c:954 Send
>>>>>>> signal
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971
>>>>>>> variable
>>>>>>> string 0 = [origination_caller_id_number=03996590031055]
>>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971
>>>>>>> variable
>>>>>>> string 1 = [ccd_session_id=20110610105829676824]
>>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971
>>>>>>> variable
>>>>>>> string 2 = [sip_cid_type=pid]
>>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_ivr_originate.c:1971
>>>>>>> variable
>>>>>>> string 3 = [privacy=yes]
>>>>>>> 2011-06-10 11:10:27.266218 [NOTICE] switch_channel.c:808 New Channel
>>>>>>> sofia/external/03996563750914 at 203.208.207.212
>>>>>>> [30228d2b-756a-4a98-871d-db63a2955b52]
>>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] mod_sofia.c:4129
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_NEW
>>>>>>> ->
>>>>>>> CS_INIT
>>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_session.c:1116 Send
>>>>>>> signal
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:320
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>>>> CS_INIT
>>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:356
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State INIT
>>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] mod_sofia.c:84
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA INIT
>>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] mod_sofia.c:124
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change CS_INIT
>>>>>>> ->
>>>>>>> CS_ROUTING
>>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_session.c:1116 Send
>>>>>>> signal
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:356
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State INIT going to
>>>>>>> sleep
>>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:320
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>>>> CS_ROUTING
>>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_channel.c:1657
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Callstate Change
>>>>>>> DOWN ->
>>>>>>> RINGING
>>>>>>> 2011-06-10 11:10:27.266218 [DEBUG] switch_core_state_machine.c:359
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State ROUTING
>>>>>>> send 984 bytes to udp/[203.208.207.212]:5060 at 03:10:27.373220:
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>>   INVITE sip:03996563750914 at 203.208.207.212 SIP/2.0
>>>>>>>   Via: SIP/2.0/UDP
>>>>>>> 202.73.56.46:5080;rport;branch=z9hG4bKZ49rj5Qtr8aBN2011-06-10
>>>>>>> 11:10:27.266218 [DEBUG] mod_sofia.c:147
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA ROUTING
>>>>>>>
>>>>>>>   Max-Forwards: 70
>>>>>>>   From: "" &lt;sip:03996590031055 at 202.73.56.46&gt;;tag=jDSe6ycN62Zar
>>>>>>>   To: &lt;sip:03996563750914 at 203.208.207.212&gt;
>>>>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>>>>   CSeq: 13501649 INVITE
>>>>>>>   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
>>>>>>>   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
>>>>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>>>>> REGISTER, REFER, NOTIFY
>>>>>>>   Supported: timer, precondition, path, replaces
>>>>>>>   Allow-Events: talk, hold, refer
>>>>>>>   Privacy: none
>>>>>>>   Content-Type: application/sdp
>>>>>>>   Content-Disposition: session
>>>>>>>   Content-Length: 204
>>>>>>>   X-FS-Support: update_display
>>>>>>>   P-Asserted-Identity: "" &lt;sip:03996590031055 at 202.73.56.46&gt;
>>>>>>>
>>>>>>>   v=0
>>>>>>>   o=FreeSWITCH 1307646863 1307646864 IN IP4 202.73.56.46
>>>>>>>   s=FreeSWITCH
>>>>>>>   c=IN IP4 202.73.56.46
>>>>>>>   t=0 0
>>>>>>>   m=audio 28564 RTP/AVP 18 3 101 13
>>>>>>>   a=rtpmap:101 telephone-event/8000
>>>>>>>   a=fmtp:101 0-16
>>>>>>>   a=ptime:20
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>> 2011-06-10 11:10:27.267630 [DEBUG] sofia.c:4646 Channel
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 entering state
>>>>>>> [calling][0]
>>>>>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_ivr_originate.c:66
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change
>>>>>>> CS_ROUTING ->
>>>>>>> CS_CONSUME_MEDIA
>>>>>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_session.c:1116 Send
>>>>>>> signal
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:359
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State ROUTING going
>>>>>>> to sleep
>>>>>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:320
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>>>> CS_CONSUME_MEDIA
>>>>>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:378
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State CONSUME_MEDIA
>>>>>>> 2011-06-10 11:10:27.267630 [DEBUG] switch_core_state_machine.c:378
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State CONSUME_MEDIA
>>>>>>> going to
>>>>>>> sleep
>>>>>>> recv 307 bytes from udp/[203.208.207.212]:5060 at 03:10:27.378710:
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>>   SIP/2.0 100 Trying
>>>>>>>   Via: SIP/2.0/UDP
>>>>>>> 10.1.1.46:5080
>>>>>>> ;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>>>>>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>>>>>>   To: &lt;sip:03996563750914 at 203.208.207.212&gt;
>>>>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>>>>   CSeq: 13501649 INVITE
>>>>>>>   Content-Length:     0
>>>>>>>
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>> 2011-06-10 11:10:27.278484 [DEBUG] switch_ivr.c:563
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 Command Execute
>>>>>>> playback(/usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav)
>>>>>>> EXECUTE sofia/external/03996590031055 at 203.208.207.212
>>>>>>> playback(/usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav)
>>>>>>> 2011-06-10 11:10:27.278484 [DEBUG] switch_core_file.c:176 File
>>>>>>> /usr/local/freeswitch/sounds/clearhub/MusicForCalls_v1.wav sample
>>>>>>> rate 11025
>>>>>>> doesn't match requested rate 8000
>>>>>>> 2011-06-10 11:10:27.278484 [WARNING] switch_core_file.c:189 File has
>>>>>>> 2
>>>>>>> channels, muxing to mono will occur.
>>>>>>> 2011-06-10 11:10:27.278484 [DEBUG] switch_ivr_play_say.c:1244 Codec
>>>>>>> Activated L16 at 8000hz 2 channels 20ms
>>>>>>> 2011-06-10 11:10:27.298764 [INFO] mod_com_g729.c:119 ENCODER CREATE -
>>>>>>> 0x2aaab00310c0 0x2aaab00b20c0
>>>>>>> recv 666 bytes from udp/[203.208.207.212]:5060 at 03:10:28.618472:
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>>   SIP/2.0 183 Session Progress
>>>>>>>   Via: SIP/2.0/UDP
>>>>>>> 10.1.1.46:5080
>>>>>>> ;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>>>>>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>>>>>>   To:
>>>>>>> &lt;sip:03996563750914 at 203.208.207.212
>>>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
>>>>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>>>>   CSeq: 13501649 INVITE
>>>>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER,
>>>>>>> NOTIFY,
>>>>>>> SUBSCRIBE, UPDATE
>>>>>>>   Content-Type: application/sdp
>>>>>>>   Content-Length:   186
>>>>>>>
>>>>>>>   v=0
>>>>>>>   o=- 131082 1 IN IP4 203.208.207.218
>>>>>>>   s=session
>>>>>>>   c=IN IP4 203.208.207.195
>>>>>>>   t=0 0
>>>>>>>   m=audio 45002 RTP/AVP 18 101
>>>>>>>   a=rtpmap:18 G729/8000
>>>>>>>   a=rtpmap:101 telephone-event/8000
>>>>>>>   a=sendrecv
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>> 2011-06-10 11:10:28.513195 [INFO] sofia.c:729
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 Update Callee ID to
>>>>>>> "Outbound
>>>>>>> Call" <03996563750914>
>>>>>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia.c:4646 Channel
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 entering state
>>>>>>> [proceeding][183]
>>>>>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia.c:4657 Remote SDP:
>>>>>>> v=0
>>>>>>> o=- 131082 1 IN IP4 203.208.207.218
>>>>>>> s=session
>>>>>>> c=IN IP4 203.208.207.195
>>>>>>> t=0 0
>>>>>>> m=audio 45002 RTP/AVP 18 101
>>>>>>> a=rtpmap:18 G729/8000
>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>
>>>>>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:4467 Audio Codec
>>>>>>> Compare
>>>>>>> [G729:18:8000:20:8000]/[G729:18:8000:20:8000]
>>>>>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:2757 Set Codec
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 G729/8000 20 ms 160
>>>>>>> samples
>>>>>>> 8000 bits
>>>>>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:4565 Set 2833 dtmf
>>>>>>> send
>>>>>>> payload to 101
>>>>>>> 2011-06-10 11:10:28.513195 [DEBUG] sofia_glue.c:2987 AUDIO RTP
>>>>>>> [sofia/external/03996563750914 at 203.208.207.212] 10.1.1.46 port 28564
>>>>>>> ->
>>>>>>> 203.208.207.195 port 45002 codec: 18 ms: 20
>>>>>>> 2011-06-10 11:10:28.513195 [DEBUG] switch_rtp.c:1607 Starting timer
>>>>>>> [soft]
>>>>>>> 160 bytes per 20ms
>>>>>>> 2011-06-10 11:10:28.514938 [DEBUG] sofia_glue.c:3228 Set 2833 dtmf
>>>>>>> send
>>>>>>> payload to 101
>>>>>>> 2011-06-10 11:10:28.514938 [DEBUG] sofia_glue.c:3233 Set 2833 dtmf
>>>>>>> receive
>>>>>>> payload to 101
>>>>>>> 2011-06-10 11:10:28.514938 [NOTICE] sofia_glue.c:3680 Pre-Answer
>>>>>>> sofia/external/03996563750914 at 203.208.207.212!
>>>>>>> 2011-06-10 11:10:28.514938 [DEBUG] switch_channel.c:2627
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Callstate Change
>>>>>>> RINGING ->
>>>>>>> EARLY
>>>>>>> 2011-06-10 11:10:28.514938 [DEBUG] switch_ivr_originate.c:3408
>>>>>>> Originate
>>>>>>> Resulted in Success: [sofia/external/03996563750914 at 203.208.207.212]
>>>>>>> 2011-06-10 11:10:28.514938 [DEBUG] mod_commands.c:3205
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change
>>>>>>> CS_CONSUME_MEDIA -> CS_EXECUTE
>>>>>>> 2011-06-10 11:10:28.514938 [DEBUG] switch_core_session.c:1116 Send
>>>>>>> signal
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>>>> 2011-06-10 11:10:28.515884 [DEBUG] switch_core_state_machine.c:320
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>>>> CS_EXECUTE
>>>>>>> 2011-06-10 11:10:28.515884 [DEBUG] switch_core_state_machine.c:366
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State EXECUTE
>>>>>>> 2011-06-10 11:10:28.515884 [DEBUG] mod_sofia.c:240
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA EXECUTE
>>>>>>> 2011-06-10 11:10:28.515884 [DEBUG] switch_core_state_machine.c:157
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 Standard EXECUTE
>>>>>>> EXECUTE sofia/external/03996563750914 at 203.208.207.212 park()
>>>>>>> 2011-06-10 11:10:28.516887 [DEBUG] switch_core_session.c:954 Send
>>>>>>> signal
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_ivr_bridge.c:1480
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change
>>>>>>> CS_EXECUTE ->
>>>>>>> CS_HIBERNATE
>>>>>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:1116 Send
>>>>>>> signal
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_ivr_bridge.c:1482
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change
>>>>>>> CS_EXECUTE ->
>>>>>>> CS_HIBERNATE
>>>>>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:1116 Send
>>>>>>> signal
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:771 Send
>>>>>>> signal
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>>>> 2011-06-10 11:10:28.517896 [DEBUG] switch_core_session.c:771 Send
>>>>>>> signal
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_ivr_play_say.c:1581 done
>>>>>>> playing
>>>>>>> file
>>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_ivr.c:563
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 Command Execute
>>>>>>> playback(tone_stream://%(2000,4000,440,480);loops=10)
>>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:366
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State EXECUTE going
>>>>>>> to sleep
>>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:320
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>>>> CS_HIBERNATE
>>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:381
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State HIBERNATE
>>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] mod_sofia.c:221
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA HIBERNATE
>>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_ivr_bridge.c:731
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change
>>>>>>> CS_HIBERNATE ->
>>>>>>> CS_RESET
>>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_session.c:1116 Send
>>>>>>> signal
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:381
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State HIBERNATE
>>>>>>> going to
>>>>>>> sleep
>>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:320
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>>>> CS_RESET
>>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:362
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State RESET
>>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] mod_sofia.c:165
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA RESET
>>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_ivr_bridge.c:716
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 CUSTOM RESET
>>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:66
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 Standard RESET
>>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_state_machine.c:362
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State RESET going to
>>>>>>> sleep
>>>>>>> 2011-06-10 11:10:28.519223 [DEBUG] switch_core_session.c:709 Send
>>>>>>> signal
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>>>> recv 666 bytes from udp/[203.208.207.212]:5060 at 03:10:30.540814:
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>>   SIP/2.0 183 Session Progress
>>>>>>>   Via: SIP/2.0/UDP
>>>>>>> 10.1.1.46:5080
>>>>>>> ;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>>>>>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>>>>>>   To:
>>>>>>> &lt;sip:03996563750914 at 203.208.207.212
>>>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
>>>>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>>>>   CSeq: 13501649 INVITE
>>>>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER,
>>>>>>> NOTIFY,
>>>>>>> SUBSCRIBE, UPDATE
>>>>>>>   Content-Type: application/sdp
>>>>>>>   Content-Length:   186
>>>>>>>
>>>>>>>   v=0
>>>>>>>   o=- 131082 2 IN IP4 203.208.207.218
>>>>>>>   s=session
>>>>>>>   c=IN IP4 203.208.207.195
>>>>>>>   t=0 0
>>>>>>>   m=audio 45002 RTP/AVP 18 101
>>>>>>>   a=rtpmap:18 G729/8000
>>>>>>>   a=rtpmap:101 telephone-event/8000
>>>>>>>   a=sendrecv
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>> 2011-06-10 11:10:30.435309 [INFO] sofia.c:729
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 Update Callee ID to
>>>>>>> "03996563750914" <03996563750914>
>>>>>>> 2011-06-10 11:10:30.435309 [DEBUG] sofia.c:4641 Channel
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 skipping state
>>>>>>> [proceeding][183]
>>>>>>> 2011-06-10 11:10:30.691404 [WARNING] switch_core_session.c:1940
>>>>>>> Cannot
>>>>>>> execute app 'playback' media required on an outbound channel that
>>>>>>> does not
>>>>>>> have media established
>>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:366
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State EXECUTE going
>>>>>>> to sleep
>>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>>>> CS_HIBERNATE
>>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:381
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State HIBERNATE
>>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:221
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA HIBERNATE
>>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:731
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change
>>>>>>> CS_HIBERNATE ->
>>>>>>> CS_RESET
>>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_session.c:1116 Send
>>>>>>> signal
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:381
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State HIBERNATE
>>>>>>> going to
>>>>>>> sleep
>>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>>>> CS_RESET
>>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:362
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State RESET
>>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:165
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA RESET
>>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:716
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 CUSTOM RESET
>>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:723
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change
>>>>>>> CS_RESET ->
>>>>>>> CS_SOFT_EXECUTE
>>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_session.c:1116 Send
>>>>>>> signal
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:362
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State RESET going to
>>>>>>> sleep
>>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>>>> CS_SOFT_EXECUTE
>>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:372
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State SOFT_EXECUTE
>>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:558 SOFIA SOFT_EXECUTE
>>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:741
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 CUSTOM SOFT_EXECUTE
>>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:761
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change
>>>>>>> CS_RESET ->
>>>>>>> CS_SOFT_EXECUTE
>>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_session.c:1116 Send
>>>>>>> signal
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:320
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>>>> CS_SOFT_EXECUTE
>>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:372
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State SOFT_EXECUTE
>>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] mod_sofia.c:558 SOFIA SOFT_EXECUTE
>>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_ivr_bridge.c:741
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 CUSTOM SOFT_EXECUTE
>>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:204
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 Standard SOFT_EXECUTE
>>>>>>> 2011-06-10 11:10:30.691404 [DEBUG] switch_core_state_machine.c:372
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State SOFT_EXECUTE
>>>>>>> going to
>>>>>>> sleep
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2535
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Callstate Change
>>>>>>> EARLY ->
>>>>>>> HANGUP
>>>>>>> 2011-06-10 11:10:30.711798 [NOTICE] switch_ivr_originate.c:1045
>>>>>>> Hangup
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 [CS_SOFT_EXECUTE]
>>>>>>> [DESTINATION_OUT_OF_ORDER]
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2551 Send signal
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 [KILL]
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send
>>>>>>> signal
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>>>> CS_HANGUP
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2535
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change
>>>>>>> ACTIVE ->
>>>>>>> HANGUP
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State HANGUP
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:457 Channel
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 hanging up, cause:
>>>>>>> DESTINATION_OUT_OF_ORDER
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:510 Sending CANCEL to
>>>>>>> sofia/external/03996563750914 at 203.208.207.212
>>>>>>> send 390 bytes to udp/[203.208.207.212]:5060 at 03:10:30.818391:
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>>   CANCEL sip:03996563750914 at 203.208.207.212 SIP/2.0
>>>>>>>   Via: SIP/2.0/UDP 202.73.56.46:5080
>>>>>>> ;rport;branch=z9hG4bKZ49rj5Qtr8aBN
>>>>>>>   Max-Forwards: 70
>>>>>>>   From: "" &lt;sip:03996590031055 at 202.73.56.46&gt;;tag=jDSe6ycN62Zar
>>>>>>>   To: &lt;sip:03996563750914 at 203.208.207.212&gt;
>>>>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>>>>   CSeq: 13501649 CANCEL
>>>>>>>   Reason: Q.850;cause=27;text="DESTINATION_OUT_OF_ORDER"
>>>>>>>   Content-Length: 0
>>>>>>>
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>> 2011-06-10 11:10:30.711798 [NOTICE] switch_ivr_bridge.c:772 Hangup
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 [CS_SOFT_EXECUTE]
>>>>>>> [ORIGINATOR_CANCEL]
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:46
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 Standard HANGUP,
>>>>>>> cause:
>>>>>>> DESTINATION_OUT_OF_ORDER
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State HANGUP going
>>>>>>> to sleep
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:351
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change
>>>>>>> CS_HANGUP ->
>>>>>>> CS_REPORTING
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send
>>>>>>> signal
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>>>> CS_REPORTING
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:617
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State REPORTING
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:53
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 Standard REPORTING,
>>>>>>> cause:
>>>>>>> DESTINATION_OUT_OF_ORDER
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:617
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State REPORTING
>>>>>>> going to
>>>>>>> sleep
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_channel.c:2551 Send signal
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 [KILL]
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send
>>>>>>> signal
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:372
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State SOFT_EXECUTE
>>>>>>> going to
>>>>>>> sleep
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>>>> CS_HANGUP
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:345
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State Change
>>>>>>> CS_REPORTING ->
>>>>>>> CS_DESTROY
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send
>>>>>>> signal
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 [BREAK]
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1288 Session
>>>>>>> 40
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Locked, Waiting on
>>>>>>> external
>>>>>>> entities
>>>>>>> 2011-06-10 11:10:30.711798 [NOTICE] switch_core_session.c:1306
>>>>>>> Session 40
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Ended
>>>>>>> 2011-06-10 11:10:30.711798 [NOTICE] switch_core_session.c:1308 Close
>>>>>>> Channel
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 [CS_DESTROY]
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:449
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Callstate Change
>>>>>>> HANGUP ->
>>>>>>> DOWN
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:452
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) Running State Change
>>>>>>> CS_DESTROY
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:462
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State DESTROY
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:362
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 SOFIA DESTROY
>>>>>>> 2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:78 ENCODER DESTROYX
>>>>>>> -
>>>>>>> 0x2aaaac013028 (nil)
>>>>>>> 2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:79 DECODER DESTROYX
>>>>>>> -
>>>>>>> 0x2aaaac013028 (nil)
>>>>>>> 2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:78 ENCODER DESTROYX
>>>>>>> -
>>>>>>> 0x2aaaac013088 (nil)
>>>>>>> 2011-06-10 11:10:30.711798 [INFO] mod_com_g729.c:79 DECODER DESTROYX
>>>>>>> -
>>>>>>> 0x2aaaac013088 (nil)
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:60
>>>>>>> sofia/external/03996563750914 at 203.208.207.212 Standard DESTROY
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:462
>>>>>>> (sofia/external/03996563750914 at 203.208.207.212) State DESTROY going
>>>>>>> to sleep
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State HANGUP
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:457 Channel
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 hanging up, cause:
>>>>>>> ORIGINATOR_CANCEL
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] mod_sofia.c:500 Sending BYE to
>>>>>>> sofia/external/03996590031055 at 203.208.207.212
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:46
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 Standard HANGUP,
>>>>>>> cause:
>>>>>>> ORIGINATOR_CANCEL
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:557
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State HANGUP going
>>>>>>> to sleep
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:351
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change
>>>>>>> CS_HANGUP ->
>>>>>>> CS_REPORTING
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_session.c:1116 Send
>>>>>>> signal
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:320
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>>>> CS_REPORTING
>>>>>>> 2011-06-10 11:10:30.711798 [DEBUG] switch_core_state_machine.c:617
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State REPORTING
>>>>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:53
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 Standard REPORTING,
>>>>>>> cause:
>>>>>>> ORIGINATOR_CANCEL
>>>>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:617
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State REPORTING
>>>>>>> going to
>>>>>>> sleep
>>>>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:345
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State Change
>>>>>>> CS_REPORTING ->
>>>>>>> CS_DESTROY
>>>>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_session.c:1116 Send
>>>>>>> signal
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 [BREAK]
>>>>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_session.c:1288 Session
>>>>>>> 39
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Locked, Waiting on
>>>>>>> external
>>>>>>> entities
>>>>>>> 2011-06-10 11:10:30.714001 [NOTICE] switch_core_session.c:1306
>>>>>>> Session 39
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Ended
>>>>>>> 2011-06-10 11:10:30.714001 [NOTICE] switch_core_session.c:1308 Close
>>>>>>> Channel
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 [CS_DESTROY]
>>>>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:449
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Callstate Change
>>>>>>> HANGUP ->
>>>>>>> DOWN
>>>>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:452
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) Running State Change
>>>>>>> CS_DESTROY
>>>>>>> 2011-06-10 11:10:30.714001 [DEBUG] switch_core_state_machine.c:462
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State DESTROY
>>>>>>> 2011-06-10 11:10:30.714001 [DEBUG] mod_sofia.c:362
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 SOFIA DESTROY
>>>>>>> 2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:78 ENCODER DESTROYX
>>>>>>> -
>>>>>>> 0x2aaab0031060 (nil)
>>>>>>> 2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:79 DECODER DESTROYX
>>>>>>> -
>>>>>>> 0x2aaab0031060 (nil)
>>>>>>> 2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:78 ENCODER DESTROYX
>>>>>>> -
>>>>>>> 0x2aaab00310c0 0x2aaab00b20c0
>>>>>>> 2011-06-10 11:10:30.714001 [INFO] mod_com_g729.c:79 DECODER DESTROYX
>>>>>>> -
>>>>>>> 0x2aaab00310c0 (nil)
>>>>>>> send 662 bytes to udp/[203.208.207.212]:5060 at 03:10:30.820530:
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>>   BYE sip:203.208.207.212:5060 SIP/2.0
>>>>>>>   Via: SIP/2.0/UDP 202.73.56.46:5080
>>>>>>> ;rport;branch=z9hG4bK0D3Hm08XNH1Xg
>>>>>>>   Max-Forwards: 70
>>>>>>>   From: "" &lt;sip:03996563750914 at 202.73.56.46&gt;;tag=H4ZN43UH9S9Qc
>>>>>>>   To:
>>>>>>> &lt;sip:03996590031055 at 203.208.207.212
>>>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>>>>   CSeq: 13501634 BYE
>>>>>>>   Contact: &lt;sip:mod_sofia at 202.73.56.46:5080&gt;
>>>>>>>   User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110603T012235Z
>>>>>>>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>>>>> REGISTER, REFER, NOTIFY
>>>>>>>   Supported: timer, precondition, path, replaces
>>>>>>>   Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL"
>>>>>>>   Content-Length: 0
>>>>>>>
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>> 2011-06-10 11:10:30.715878 [INFO] mod_com_g729.c:83 ENCODER DESTROY -
>>>>>>> 0x2aaab00310c0 0x2aaab00b20c0
>>>>>>> 2011-06-10 11:10:30.715878 [DEBUG] switch_core_state_machine.c:60
>>>>>>> sofia/external/03996590031055 at 203.208.207.212 Standard DESTROY
>>>>>>> 2011-06-10 11:10:30.715878 [DEBUG] switch_core_state_machine.c:462
>>>>>>> (sofia/external/03996590031055 at 203.208.207.212) State DESTROY going
>>>>>>> to sleep
>>>>>>> recv 383 bytes from udp/[203.208.207.212]:5060 at 03:10:30.823302:
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>>   SIP/2.0 200 Ok
>>>>>>>   Via: SIP/2.0/UDP
>>>>>>> 10.1.1.46:5080
>>>>>>> ;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>>>>>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>>>>>>   To:
>>>>>>> &lt;sip:03996563750914 at 203.208.207.212
>>>>>>> &gt;;tag=2QGB951HCR30000E1D00000u00000001QXU3LU
>>>>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>>>>   CSeq: 13501649 CANCEL
>>>>>>>   Contact: <sip:203.208.207.212:5060>
>>>>>>>   Content-Length:     0
>>>>>>>
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>> recv 411 bytes from udp/[203.208.207.212]:5060 at 03:10:30.824765:
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>>   SIP/2.0 487 Request Terminated
>>>>>>>   Via: SIP/2.0/UDP
>>>>>>> 10.1.1.46:5080
>>>>>>> ;rport=5080;branch=z9hG4bKZ49rj5Qtr8aBN;received=10.1.1.46
>>>>>>>   From: "" &lt;sip:03996590031055 at 10.1.1.46&gt;;tag=jDSe6ycN62Zar
>>>>>>>   To:
>>>>>>> &lt;sip:03996563750914 at 203.208.207.212
>>>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
>>>>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>>>>   CSeq: 13501649 INVITE
>>>>>>>   Reason: SIP;cause=487;text="Request Terminated"
>>>>>>>   Content-Length:     0
>>>>>>>
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>> send 371 bytes to udp/[203.208.207.212]:5060 at 03:10:30.824891:
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>>   ACK sip:03996563750914 at 203.208.207.212 SIP/2.0
>>>>>>>   Via: SIP/2.0/UDP 202.73.56.46:5080
>>>>>>> ;rport;branch=z9hG4bKZ49rj5Qtr8aBN
>>>>>>>   Max-Forwards: 70
>>>>>>>   From: "" &lt;sip:03996590031055 at 202.73.56.46&gt;;tag=jDSe6ycN62Zar
>>>>>>>   To:
>>>>>>> &lt;sip:03996563750914 at 203.208.207.212
>>>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ410OWC5F
>>>>>>>   Call-ID: 082c8244-0db2-122f-c4ab-0015c5fc7ad9
>>>>>>>   CSeq: 13501649 ACK
>>>>>>>   Content-Length: 0
>>>>>>>
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>> recv 380 bytes from udp/[203.208.207.212]:5060 at 03:10:30.826375:
>>>>>>>
>>>>>>> ------------------------------------------------------------------------
>>>>>>>   SIP/2.0 200 Ok
>>>>>>>   Via: SIP/2.0/UDP
>>>>>>> 10.1.1.46:5080
>>>>>>> ;rport=5080;branch=z9hG4bK0D3Hm08XNH1Xg;received=10.1.1.46
>>>>>>>   From: "" &lt;sip:03996563750914 at 10.1.1.46&gt;;tag=H4ZN43UH9S9Qc
>>>>>>>   To:
>>>>>>> &lt;sip:03996590031055 at 203.208.207.212
>>>>>>> &gt;;tag=2QGB951HCR30000E1D0001Ll0008FJ21EI9PBW
>>>>>>>   Call-ID: f529ae7e-0db1-122f-c4ab-0015c5fc7ad9
>>>>>>>   CSeq: 13501634 BYE
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> View this message in context:
>>>>>>> http://freeswitch-users.2379917.n2.nabble.com/Leg-A-is-automatically-disconnected-on-Leg-B-orginate-failure-tp6482192p6482192.html
>>>>>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
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From anthony.minessale at gmail.com  Tue Jun 28 19:57:15 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 28 Jun 2011 10:57:15 -0500
Subject: [Freeswitch-users] Copy custom SIP headers -
 {sip_copy_custom_headers=true} doesn't seem to be working
In-Reply-To: <BANLkTikNA_oA_yswO+7KZ5ir7YUVDNjM2A@mail.gmail.com>
References: <BANLkTikNA_oA_yswO+7KZ5ir7YUVDNjM2A@mail.gmail.com>
Message-ID: <BANLkTimR7EAhpxaX-hRe128tq4VEF=yoeQ@mail.gmail.com>

Problem seems to be we did not support final responses only 1xx and
200. I think I missed that part of the other guy's report.
Update again to lastest GIT and it should work.

This is one reason we need JIRA for issues of this type because it's
easy to lose track of email threads in a big list like this.



On Tue, Jun 28, 2011 at 2:19 AM, Jonas Borjesson
<jonas at jonasborjesson.com> wrote:
> Hi all,
>
> As many others, I need to copy customer X-headers from leg B back over
> to leg A and have found a few threads talking about using the
> sip_copy_custom_headers=true with the dial string but this does not
> seem to be working. I am doing exactly what is described in the thread
> below but like that guy, it does not work for me either and that
> thread never came to any conclusion. Does anyone know the status of
> this feature or what I may be doing wrong? Any help is greatly
> appreciated!
>
> Of course, I tried on the latest and greatest FS -
> 1.0.head-git-4962542 2011-06-27 10-15-03 -0500
>
> The tread talking about this very feature:
> http://comments.gmane.org/gmane.comp.telephony.freeswitch.user/36287
>
> And documentation:
> http://wiki.freeswitch.org/wiki/Variable_sip_copy_custom_headers
>
> Thanks,
>
> /Jonas
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From steveu at coppice.org  Tue Jun 28 19:58:11 2011
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 28 Jun 2011 23:58:11 +0800
Subject: [Freeswitch-users] Proper prompt gain/level
In-Reply-To: <2D33125A-05E0-40DE-92A2-7038940977AB@bryansmart.com>
References: <FE081E27-8D62-4715-B7AA-64432598E013@bryansmart.com>	<0D3AEFF37C1643B79CE3F7A7EE9EAEC3@dell9400>	<66E81300-46E9-4C81-B2E0-123BCADDCAEB@bryansmart.com>	<4E088A24.7010002@coppice.org>	<3081AB39-6EF0-4C43-9AC1-7D43FCD378CD@bryansmart.com>	<4E092ED4.60605@coppice.org>
	<2D33125A-05E0-40DE-92A2-7038940977AB@bryansmart.com>
Message-ID: <4E09FA13.5030104@coppice.org>

On 06/28/2011 11:04 AM, Bryan Smart wrote:
> I think that dBm0 only applies if we are measuring the power on an analog circuit, or at the d/a point of a digital circuit.
Nope.
> I was performing analysis of a digitally encoded audio file, therefore the measurement is dBFS, and 0DBFS is the point where clipping takes place. Anything below 0DBFS does not clip, even though it might seem loud to someone.
What is your notion if dBFS? There are two - one where 0dBFS is a sine 
wave touching the limits of the number range, and the other where 0dBFS 
is a square wave touching those limits. They are only 3dB apart, but 
dbOv and dBm0 are better defined scales.

Your notion is 0dBFS not clipping is based on a sine wave. For speech 
you need something lower because of the statistics of speech.
> I stated that the peak power of the stock prompts are typically -15 to -16 (DBFS), and the short term RMS power is about -32DBFS.
That statement doesn't seem to make much sense.
> I don't know how to properly evaluate how DBFS will convert to DBm0. I gather that the codec and the d/a converter attenuate the level to some extent, but I don't know how much. If I have a digital file with a 1Khz test tone at -10DBFS, and play that over the pstn, what do I get out in DBm0?
A sine wave touching the limits of the digital range is about +3.14dBm0.
> Speaking just in terms of DBFS, -16DB is only about 15% of the potential power available before the audio reaches 0DB, and clips. Perhaps when -16DBFS is put out over the pstn, it is much closer to clipping than it would be in an entirely digital domain. I don't know enough to make that determination. Do you know?
Yes, and you should know, since I told you last time.
> I'm not for clipping and distortion. However, too little power can cause another problem: limited dynamic range and increased dithering artifacts. If audio is quiet on a pstn phone, then the person with the phone might be able to increase the level by turning up the phone's volume, if it has one. However, that raises the noise floor. Companding might make 8-bit channels sound a bit like 14-bit or 15-bit channels in terms of a low noise floor and decreased dithering artifacts, but it is still just an 8-bit channel. Companding hides most of the dithering artifacts in strong signals, but it magnifies them in weak signals. That's why G711 is fairly clear, but will sound scratchy if you put faint signals in to it and try to amplify them back up to normal levels. Over G711 or any pstn call, as you decrease the level of the audio going across, the scratchy dithering static obscures an increasing amount of the audio.
That paragraph is utter drivel. The companding of G.711 ensures that the 
audio quality is roughly maintained from clipping down to -45dBm0 or so. 
Below that the quality falls, just like most other digital coding scheme 
running out of bits. The FS prompts aren't nearly that quiet.
> This gets worse if you compound it by stacking codecs. Consider if someone calls your IVR from a cell phone, and you play quiet prompts to them. The audio is first passed through G711, where the low gain means that scratchy dithering artifacts are added. Then, it is encoded to GSM for the cell phone,. GSM uses linear predictive coding, and, being tuned for voice, it is not optimized for having smooth waveforms interrupted periodically with random excursions. Thats part of why cell phone calls sound so bad if there is lots of background noise.
Again, this is drivel.
> Anyway, clipping is to be avoided, but simply reducing levels dramatically creates other quality problems on a channel that uses companding. You trade distorted audio for scratchy audio. Same thing happened with cassette tapes that used Dolby noise reduction.
You really are just making this up as you go along, aren't you.
> Maybe you can help clear up my understanding of how DBFS in a digital file will work out in DBm0 on a pstn line. As things stand, though, I think we have lots of room to increase the level of the prompts before we reach a point where clipping is an issue.
I tried playing with a few of the prompts. The peaks of the words are 
about -20dBm0, or -26dBOv. That's probably -23 or -26 on your scale. 
Every word peaks to a similar level, as prompts are spoken in a pretty 
flat voice. They seem to be about 15dB away clipping, which would mean 
the crest factor is about 11dB for the Callie voice, which seems a 
little low for speech. You usually need to ride a little more than 13dB 
above the short term power of most voices to bring the crossing rate 
close to zero.

So, there is about 15dB of headroom which could be used to increase the 
speech level without clipping. Whether doing that is a good idea is 
questionable. The prompts are at about normal speech level. In normal 
speech the codecs have plenty of headroom to cope with people getting 
agitated and shouting, producing only modest amounts of clipping. Is 
cranking all the prompts up to shouting level appropriate?

Steve




From anthony.minessale at gmail.com  Tue Jun 28 20:08:50 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 28 Jun 2011 11:08:50 -0500
Subject: [Freeswitch-users] Leg-A is automatically disconnected on Leg-B
 orginate failure
In-Reply-To: <BANLkTimiExLew_1cjMuQVpF3CKkR8mrGXQ@mail.gmail.com>
References: <1308213151543-6482192.post@n2.nabble.com>
	<BANLkTi=xYgBTPVGxVGmk9f2=bmX5X2vF=Q@mail.gmail.com>
	<BANLkTikhPmjN5HeeJHBnQt1ivA8zbtCnYA@mail.gmail.com>
	<BANLkTi=m7oMix1ZRC9URtRuAbsG2fWyhMQ@mail.gmail.com>
	<BANLkTimxBU8m311LVKMWcFOjukwLg+ZUGQ@mail.gmail.com>
	<BANLkTinZJf3BxB7QHefnmOvxQmVW2r4dTg@mail.gmail.com>
	<BANLkTimiExLew_1cjMuQVpF3CKkR8mrGXQ@mail.gmail.com>
Message-ID: <BANLkTi=ZvOZ99qVxUeJAtpjtwP5r=hfVkg@mail.gmail.com>

The cases where you have this problem arise because you are
originating to the B leg who answers 183 (early media) then hangs up
before the call was answered.

set the channel variable uuid_bridge_continue_on_cancel=true on the A
leg to change the behavior to what you want.


On Mon, Jun 27, 2011 at 10:09 AM, Michael Collins <msc at freeswitch.org> wrote:
> This makes my eyes bleed. Can you please put this on
> pastebin.freeswitch.org? Use "FreeSWITCH Log" as the syntax highlight.
> -MC
>
> On Sun, Jun 26, 2011 at 11:27 PM, David Ma <mays.david at gmail.com> wrote:
>>
>> Hi Michael,
>>

-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Tue Jun 28 21:03:39 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 28 Jun 2011 12:03:39 -0500
Subject: [Freeswitch-users] ESL Send MESSAGE_WAITING Error - Cannot find
	profile
In-Reply-To: <BANLkTimGcbOiDUDgkRxbHZyPYASTidbVjQ@mail.gmail.com>
References: <BANLkTimGcbOiDUDgkRxbHZyPYASTidbVjQ@mail.gmail.com>
Message-ID: <BANLkTikHzwA0KGyBXAXxOwNid1DfMZVMKA@mail.gmail.com>

can you try this on latest too?
The line numbers suggest an older build.


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



-- 
Anthony Minessale II

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

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

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


From steveayre at gmail.com  Tue Jun 28 22:46:40 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 28 Jun 2011 19:46:40 +0100
Subject: [Freeswitch-users] UniMRCP Server as daemon won't connect RTSP
In-Reply-To: <361E98F99D3CC3439EED59BC1924ED6950815F@SERVER2003.SecuReachSystems.local>
References: <361E98F99D3CC3439EED59BC1924ED6950815F@SERVER2003.SecuReachSystems.local>
Message-ID: <EE20009F-717C-4612-9E6C-612DE77AA8CA@gmail.com>

Have you tried doing a strace or gcore of it when it's running with --daemon?

If it's running but not listening/responding it could be sonething like a locking problem. Tracing might show you what it is (or isnt) doing. Pid file could be one culprit.

Steve on iPhone



On 27 Jun 2011, at 22:52, "Jason Moran" <jmoran at secureachsystems.com> wrote:

> I?ve been playing around with IVONA?s new unimrcp plugin with unimrcp 1.0.0 (r1725) on OpenSuse 11.4
>  
> Anyway, when I start it up with the following, it stays in the foreground of my SSH w/ the following command:
> ./unimrcpserver
> That works! However, since it?s in the foreground when I close the SSH window, it kills the process.
>  
> If I use the so-called daemon mode (-d or --daemon) it says it is going into daemon mode, but FS will immediately return that it ?Failed to Connect to RTSP Server...? at the IP:port I specified. The unimrcpserver process is running, but doesn?t seem to respond to anything.
> ./unimrcpserver ?d
> ./unimrcpserver --daemon
>  
> If I attempt to background it by either using ?&? or ctrl-z it says [1]+  Stopped  ./unimrcpserver and FS will successfully make the RTSP connection but then nothing will happen. Mod_unimrcp will spit out a warning about ?MRCP session has not opened after 5000 ms?
> ./unimrcpserver &
> Those are hard to kill. But when I kill -9 it then FS will finally remove the MRCP handle and tell me that it couldn?t allocate the speech engine.
>  
> Lastly, I can nohup it, but then unimrcpserver eats up 95%+ of the CPU (instead of 1-3% as it does when I have it in the foreground), which it does not do when it runs in any other modes. It also makes a very, very large nohup.out file that keeps on growing. Even if I tell it to not log.
> nohup ./unimrcpserver &
> nohup ./unimrcpserver ?o 0 &
>  
> Ideas??
> Thanks,
> Jason Moran
>  
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From wes-fs at 499x.com  Tue Jun 28 22:57:11 2011
From: wes-fs at 499x.com (Wes)
Date: Tue, 28 Jun 2011 13:57:11 -0500
Subject: [Freeswitch-users] pause and restart while recording?
Message-ID: <4E0A2407.3010704@499x.com>

I couldn't find much on this:  Is it possible in lua script to let the 
user press a key to pause recording, and press a key again to continue 
the recording, just like a traditional recorder?

Thanks!


From bryansmart at bryansmart.com  Tue Jun 28 23:26:12 2011
From: bryansmart at bryansmart.com (Bryan Smart)
Date: Tue, 28 Jun 2011 15:26:12 -0400
Subject: [Freeswitch-users] Proper prompt gain/level
In-Reply-To: <4E09FA13.5030104@coppice.org>
References: <FE081E27-8D62-4715-B7AA-64432598E013@bryansmart.com>
	<0D3AEFF37C1643B79CE3F7A7EE9EAEC3@dell9400>
	<66E81300-46E9-4C81-B2E0-123BCADDCAEB@bryansmart.com>
	<4E088A24.7010002@coppice.org>
	<3081AB39-6EF0-4C43-9AC1-7D43FCD378CD@bryansmart.com>
	<4E092ED4.60605@coppice.org>
	<2D33125A-05E0-40DE-92A2-7038940977AB@bryansmart.com>
	<4E09FA13.5030104@coppice.org>
Message-ID: <88F60C8D-AFB1-46DB-9A88-4AB295A0376B@bryansmart.com>

I must be wrong about DBm0, then. I wasn't familiar with it, so I checked Wikipedia. Wikipedia isn't always right, of course.

Wikipedia says...
----------
dBm0 is an abbreviation for the power in dBm measured at a zero transmission level point.

dBm0 is a concept used (amongst other areas) in audio/telephony processing since it allows a smooth integration of analog and digital chains. Notably, for
A-law and ?-law codecs the standards define a sequence which has a 0 dBm0 output.

......

Note 2: 0 dBm0 is often replaced by or used instead of digital milliwatt or zero transmission level point.
----------

Where is my zero level transmission point? Data from a codec is not power being transmitted, only data for reproducing power at some later time. Isn't it all up to the D/A converter at the far end to determine what the power levels yielded by decoding the data will be represented relative to. If it is a hardware SIP phone, maybe the person has the volume cranked up, or turned down. Since the analog representation of the signal starts in the D/A that feeds the handset, what is considered 0? I have no idea. It sounds like this is a scale used for calibrating a D/A that feeds a pstn circuit.

I admit feeling frustrated by this discussion. I've mixed music and mastered CDs for nearly 15 years, and I've always felt that 0DB in an entirely digital domain is a near universally understood concept. If I have a sample of 16-bit signed LPCM audio, then -32768 or 32767 represent the max amplitude that can be stored, and is what everyone that I know in pro audio calls 0DB when speaking strictly about a file, rather than a PA, in/out levels to an analog device like a tape machine, etc. If the gain of the encoded signal is boosted to a point where none of the samples are pushed beyond this range, then none of them clip. I thought that you'd think of DB in the same way, given we were talking about levels in files, but it feels like you're nit-picking or antagonizing me. It sounds like you're telling me that they can clip, even if the gain was never increased to a point where this range would be overflowed, but I have no idea how that is possible. It does not happen when I play such audio through the D/A on a computer's sound card, nor when I store it on a CD and play it back through a stereo.

I take your point about headroom. Still, I don't feel that the current level matches caller speech. If I felt that the prompts blended well, I wouldn't have even put myself through this thread. I've connected a mix of hardware SIP phones, desktop SIP clients, and iOS SIP clients to a conference, with no modifications to the audio level of the channel, and the level of people speaking to each other in the conference is significantly louder than any prompts that are played through it. Maybe *all* of the clients are pushing audio too strongly. I first thought it was something to do with the conference, but I soon realized that the prompts were quiet everywhere, not just when played in a conference.

I don't expect anything to change due to my personal preference, and I realize that a background in digital audio as applies to music and live recording doesn't mean that I'm not ignorant about many things that involve digital audio as it applies to telephony. I raised the issue here to see if I might be doing something wrong. If not, I wondered if the prompt levels are set based on some sort of standard by people that are wiser regarding the details than me? It seems, though, that there really isn't a standard, and the decision is someone else's personal preference. I'd rather that personal preferences never be a default. Since there isn't an official standard/guideline, I suppose that someone has to make a decision, and that decision will be influenced by their own preferences.

I have a few suggestions to help improve Michael's sox script, but that's where I'll leave this issue. It isn't worth a big argument when I can fix it myself.

I apologize to the list for any smoke, hints of flames, or other frustration that might have leaked through in to my posts. The sound level thing matters to me, but it is really a small thing. I enjoy Freeswitch immensely, and really appreciate everyones' efforts in producing and updating it. I've always been fascinated with phones and any type of interactive phone app, and so Asterisk, and now Freeswitch, really spark my imagination. I'm 34, and the study of phones in my early teens was my first conceptual exposure to a large network. I phreaked a bit at the time. I was fortunate (at least in one regard) to live in the US deep south, where digital switching equipment wasn't common, and so all of the old 1970's techniques weren't unavailable. That didn't last long, but it peaked my curiosity. I used VXML for a project for an employer in 2002 or so, but didn't really feel any excitement about phones, in the way that I used to, until I ran across Asterisk. Asterisk was great for the time, but I wanted to use it more for apps than a PBX, and so was quite excited to discover the different design of Freeswitch. I rarely become excited about new environments and frameworks anymore, but Freeswitch has put me back in to a fun mood of exploration and experimentation.

Bryan

On Jun 28, 2011, at 11:58 AM, Steve Underwood wrote:

> On 06/28/2011 11:04 AM, Bryan Smart wrote:
>> I think that dBm0 only applies if we are measuring the power on an analog circuit, or at the d/a point of a digital circuit.
> Nope.
>> I was performing analysis of a digitally encoded audio file, therefore the measurement is dBFS, and 0DBFS is the point where clipping takes place. Anything below 0DBFS does not clip, even though it might seem loud to someone.
> What is your notion if dBFS? There are two - one where 0dBFS is a sine 
> wave touching the limits of the number range, and the other where 0dBFS 
> is a square wave touching those limits. They are only 3dB apart, but 
> dbOv and dBm0 are better defined scales.
> 
> Your notion is 0dBFS not clipping is based on a sine wave. For speech 
> you need something lower because of the statistics of speech.
>> I stated that the peak power of the stock prompts are typically -15 to -16 (DBFS), and the short term RMS power is about -32DBFS.
> That statement doesn't seem to make much sense.
>> I don't know how to properly evaluate how DBFS will convert to DBm0. I gather that the codec and the d/a converter attenuate the level to some extent, but I don't know how much. If I have a digital file with a 1Khz test tone at -10DBFS, and play that over the pstn, what do I get out in DBm0?
> A sine wave touching the limits of the digital range is about +3.14dBm0.
>> Speaking just in terms of DBFS, -16DB is only about 15% of the potential power available before the audio reaches 0DB, and clips. Perhaps when -16DBFS is put out over the pstn, it is much closer to clipping than it would be in an entirely digital domain. I don't know enough to make that determination. Do you know?
> Yes, and you should know, since I told you last time.
>> I'm not for clipping and distortion. However, too little power can cause another problem: limited dynamic range and increased dithering artifacts. If audio is quiet on a pstn phone, then the person with the phone might be able to increase the level by turning up the phone's volume, if it has one. However, that raises the noise floor. Companding might make 8-bit channels sound a bit like 14-bit or 15-bit channels in terms of a low noise floor and decreased dithering artifacts, but it is still just an 8-bit channel. Companding hides most of the dithering artifacts in strong signals, but it magnifies them in weak signals. That's why G711 is fairly clear, but will sound scratchy if you put faint signals in to it and try to amplify them back up to normal levels. Over G711 or any pstn call, as you decrease the level of the audio going across, the scratchy dithering static obscures an increasing amount of the audio.
> That paragraph is utter drivel. The companding of G.711 ensures that the 
> audio quality is roughly maintained from clipping down to -45dBm0 or so. 
> Below that the quality falls, just like most other digital coding scheme 
> running out of bits. The FS prompts aren't nearly that quiet.
>> This gets worse if you compound it by stacking codecs. Consider if someone calls your IVR from a cell phone, and you play quiet prompts to them. The audio is first passed through G711, where the low gain means that scratchy dithering artifacts are added. Then, it is encoded to GSM for the cell phone,. GSM uses linear predictive coding, and, being tuned for voice, it is not optimized for having smooth waveforms interrupted periodically with random excursions. Thats part of why cell phone calls sound so bad if there is lots of background noise.
> Again, this is drivel.
>> Anyway, clipping is to be avoided, but simply reducing levels dramatically creates other quality problems on a channel that uses companding. You trade distorted audio for scratchy audio. Same thing happened with cassette tapes that used Dolby noise reduction.
> You really are just making this up as you go along, aren't you.
>> Maybe you can help clear up my understanding of how DBFS in a digital file will work out in DBm0 on a pstn line. As things stand, though, I think we have lots of room to increase the level of the prompts before we reach a point where clipping is an issue.
> I tried playing with a few of the prompts. The peaks of the words are 
> about -20dBm0, or -26dBOv. That's probably -23 or -26 on your scale. 
> Every word peaks to a similar level, as prompts are spoken in a pretty 
> flat voice. They seem to be about 15dB away clipping, which would mean 
> the crest factor is about 11dB for the Callie voice, which seems a 
> little low for speech. You usually need to ride a little more than 13dB 
> above the short term power of most voices to bring the crossing rate 
> close to zero.
> 
> So, there is about 15dB of headroom which could be used to increase the 
> speech level without clipping. Whether doing that is a good idea is 
> questionable. The prompts are at about normal speech level. In normal 
> speech the codecs have plenty of headroom to cope with people getting 
> agitated and shouting, producing only modest amounts of clipping. Is 
> cranking all the prompts up to shouting level appropriate?
> 
> Steve
> 
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From anthony.minessale at gmail.com  Wed Jun 29 00:42:08 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 28 Jun 2011 15:42:08 -0500
Subject: [Freeswitch-users] Proper prompt gain/level
In-Reply-To: <88F60C8D-AFB1-46DB-9A88-4AB295A0376B@bryansmart.com>
References: <FE081E27-8D62-4715-B7AA-64432598E013@bryansmart.com>
	<0D3AEFF37C1643B79CE3F7A7EE9EAEC3@dell9400>
	<66E81300-46E9-4C81-B2E0-123BCADDCAEB@bryansmart.com>
	<4E088A24.7010002@coppice.org>
	<3081AB39-6EF0-4C43-9AC1-7D43FCD378CD@bryansmart.com>
	<4E092ED4.60605@coppice.org>
	<2D33125A-05E0-40DE-92A2-7038940977AB@bryansmart.com>
	<4E09FA13.5030104@coppice.org>
	<88F60C8D-AFB1-46DB-9A88-4AB295A0376B@bryansmart.com>
Message-ID: <BANLkTi=+DQ2h1CeS8cqkpgZ+TCnrB7=idg@mail.gmail.com>

I don't think anyone is thinking of your post as flames.  Like I said,
check out the original 48k files and see what the levels are vs what
sox does to them.  I am not adverse to changing it as long as the
majority of people agree.  Steve is very pragmatic when it comes to
DSP.  I often make a fool of myself trying to find the right words to
explain myself.  I don't believe he in any way is attempting to deter
your efforts he just is a stickler for proper terminology and factual
info.



On Tue, Jun 28, 2011 at 2:26 PM, Bryan Smart <bryansmart at bryansmart.com> wrote:
> I must be wrong about DBm0, then. I wasn't familiar with it, so I checked Wikipedia. Wikipedia isn't always right, of course.
>
> Wikipedia says...
> ----------
> dBm0 is an abbreviation for the power in dBm measured at a zero transmission level point.
>
> dBm0 is a concept used (amongst other areas) in audio/telephony processing since it allows a smooth integration of analog and digital chains. Notably, for
> A-law and ?-law codecs the standards define a sequence which has a 0 dBm0 output.
>
> ......
>
> Note 2: 0 dBm0 is often replaced by or used instead of digital milliwatt or zero transmission level point.
> ----------
>
> Where is my zero level transmission point? Data from a codec is not power being transmitted, only data for reproducing power at some later time. Isn't it all up to the D/A converter at the far end to determine what the power levels yielded by decoding the data will be represented relative to. If it is a hardware SIP phone, maybe the person has the volume cranked up, or turned down. Since the analog representation of the signal starts in the D/A that feeds the handset, what is considered 0? I have no idea. It sounds like this is a scale used for calibrating a D/A that feeds a pstn circuit.
>
> I admit feeling frustrated by this discussion. I've mixed music and mastered CDs for nearly 15 years, and I've always felt that 0DB in an entirely digital domain is a near universally understood concept. If I have a sample of 16-bit signed LPCM audio, then -32768 or 32767 represent the max amplitude that can be stored, and is what everyone that I know in pro audio calls 0DB when speaking strictly about a file, rather than a PA, in/out levels to an analog device like a tape machine, etc. If the gain of the encoded signal is boosted to a point where none of the samples are pushed beyond this range, then none of them clip. I thought that you'd think of DB in the same way, given we were talking about levels in files, but it feels like you're nit-picking or antagonizing me. It sounds like you're telling me that they can clip, even if the gain was never increased to a point where this range would be overflowed, but I have no idea how that is possible. It does not happen when I play such audio through the D/A on a computer's sound card, nor when I store it on a CD and play it back through a stereo.
>
> I take your point about headroom. Still, I don't feel that the current level matches caller speech. If I felt that the prompts blended well, I wouldn't have even put myself through this thread. I've connected a mix of hardware SIP phones, desktop SIP clients, and iOS SIP clients to a conference, with no modifications to the audio level of the channel, and the level of people speaking to each other in the conference is significantly louder than any prompts that are played through it. Maybe *all* of the clients are pushing audio too strongly. I first thought it was something to do with the conference, but I soon realized that the prompts were quiet everywhere, not just when played in a conference.
>
> I don't expect anything to change due to my personal preference, and I realize that a background in digital audio as applies to music and live recording doesn't mean that I'm not ignorant about many things that involve digital audio as it applies to telephony. I raised the issue here to see if I might be doing something wrong. If not, I wondered if the prompt levels are set based on some sort of standard by people that are wiser regarding the details than me? It seems, though, that there really isn't a standard, and the decision is someone else's personal preference. I'd rather that personal preferences never be a default. Since there isn't an official standard/guideline, I suppose that someone has to make a decision, and that decision will be influenced by their own preferences.
>
> I have a few suggestions to help improve Michael's sox script, but that's where I'll leave this issue. It isn't worth a big argument when I can fix it myself.
>
> I apologize to the list for any smoke, hints of flames, or other frustration that might have leaked through in to my posts. The sound level thing matters to me, but it is really a small thing. I enjoy Freeswitch immensely, and really appreciate everyones' efforts in producing and updating it. I've always been fascinated with phones and any type of interactive phone app, and so Asterisk, and now Freeswitch, really spark my imagination. I'm 34, and the study of phones in my early teens was my first conceptual exposure to a large network. I phreaked a bit at the time. I was fortunate (at least in one regard) to live in the US deep south, where digital switching equipment wasn't common, and so all of the old 1970's techniques weren't unavailable. That didn't last long, but it peaked my curiosity. I used VXML for a project for an employer in 2002 or so, but didn't really feel any excitement about phones, in the way that I used to, until I ran across Asterisk. Asterisk was great for the time, but I wanted to use it more for apps than a PBX, and so was quite excited to discover the different design of Freeswitch. I rarely become excited about new environments and frameworks anymore, but Freeswitch has put me back in to a fun mood of exploration and experimentation.
>
> Bryan
>
> On Jun 28, 2011, at 11:58 AM, Steve Underwood wrote:
>
>> On 06/28/2011 11:04 AM, Bryan Smart wrote:
>>> I think that dBm0 only applies if we are measuring the power on an analog circuit, or at the d/a point of a digital circuit.
>> Nope.
>>> I was performing analysis of a digitally encoded audio file, therefore the measurement is dBFS, and 0DBFS is the point where clipping takes place. Anything below 0DBFS does not clip, even though it might seem loud to someone.
>> What is your notion if dBFS? There are two - one where 0dBFS is a sine
>> wave touching the limits of the number range, and the other where 0dBFS
>> is a square wave touching those limits. They are only 3dB apart, but
>> dbOv and dBm0 are better defined scales.
>>
>> Your notion is 0dBFS not clipping is based on a sine wave. For speech
>> you need something lower because of the statistics of speech.
>>> I stated that the peak power of the stock prompts are typically -15 to -16 (DBFS), and the short term RMS power is about -32DBFS.
>> That statement doesn't seem to make much sense.
>>> I don't know how to properly evaluate how DBFS will convert to DBm0. I gather that the codec and the d/a converter attenuate the level to some extent, but I don't know how much. If I have a digital file with a 1Khz test tone at -10DBFS, and play that over the pstn, what do I get out in DBm0?
>> A sine wave touching the limits of the digital range is about +3.14dBm0.
>>> Speaking just in terms of DBFS, -16DB is only about 15% of the potential power available before the audio reaches 0DB, and clips. Perhaps when -16DBFS is put out over the pstn, it is much closer to clipping than it would be in an entirely digital domain. I don't know enough to make that determination. Do you know?
>> Yes, and you should know, since I told you last time.
>>> I'm not for clipping and distortion. However, too little power can cause another problem: limited dynamic range and increased dithering artifacts. If audio is quiet on a pstn phone, then the person with the phone might be able to increase the level by turning up the phone's volume, if it has one. However, that raises the noise floor. Companding might make 8-bit channels sound a bit like 14-bit or 15-bit channels in terms of a low noise floor and decreased dithering artifacts, but it is still just an 8-bit channel. Companding hides most of the dithering artifacts in strong signals, but it magnifies them in weak signals. That's why G711 is fairly clear, but will sound scratchy if you put faint signals in to it and try to amplify them back up to normal levels. Over G711 or any pstn call, as you decrease the level of the audio going across, the scratchy dithering static obscures an increasing amount of the audio.
>> That paragraph is utter drivel. The companding of G.711 ensures that the
>> audio quality is roughly maintained from clipping down to -45dBm0 or so.
>> Below that the quality falls, just like most other digital coding scheme
>> running out of bits. The FS prompts aren't nearly that quiet.
>>> This gets worse if you compound it by stacking codecs. Consider if someone calls your IVR from a cell phone, and you play quiet prompts to them. The audio is first passed through G711, where the low gain means that scratchy dithering artifacts are added. Then, it is encoded to GSM for the cell phone,. GSM uses linear predictive coding, and, being tuned for voice, it is not optimized for having smooth waveforms interrupted periodically with random excursions. Thats part of why cell phone calls sound so bad if there is lots of background noise.
>> Again, this is drivel.
>>> Anyway, clipping is to be avoided, but simply reducing levels dramatically creates other quality problems on a channel that uses companding. You trade distorted audio for scratchy audio. Same thing happened with cassette tapes that used Dolby noise reduction.
>> You really are just making this up as you go along, aren't you.
>>> Maybe you can help clear up my understanding of how DBFS in a digital file will work out in DBm0 on a pstn line. As things stand, though, I think we have lots of room to increase the level of the prompts before we reach a point where clipping is an issue.
>> I tried playing with a few of the prompts. The peaks of the words are
>> about -20dBm0, or -26dBOv. That's probably -23 or -26 on your scale.
>> Every word peaks to a similar level, as prompts are spoken in a pretty
>> flat voice. They seem to be about 15dB away clipping, which would mean
>> the crest factor is about 11dB for the Callie voice, which seems a
>> little low for speech. You usually need to ride a little more than 13dB
>> above the short term power of most voices to bring the crossing rate
>> close to zero.
>>
>> So, there is about 15dB of headroom which could be used to increase the
>> speech level without clipping. Whether doing that is a good idea is
>> questionable. The prompts are at about normal speech level. In normal
>> speech the codecs have plenty of headroom to cope with people getting
>> agitated and shouting, producing only modest amounts of clipping. Is
>> cranking all the prompts up to shouting level appropriate?
>>
>> Steve
>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Wed Jun 29 02:34:14 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 28 Jun 2011 17:34:14 -0500
Subject: [Freeswitch-users] ClueCon Registration by July 4th Means More
	Chances to Win
Message-ID: <BANLkTim8hDQK8K=DY-EFy8NA2acTRuqdeA@mail.gmail.com>

Don't forget that anyone who registers for ClueCon before July 4th
gets 3 extra tickets to win the prizes!
Now is your chance to interact with our fast-paced world-wide
development community in person....

http://www.cluecon.com

visit our community conference for tomorrow's meeting or anytime at
http://conference.freeswitch.org and press the register for ClueCon
button to register by web phone right from your browser!


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
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From msc at freeswitch.org  Wed Jun 29 03:11:10 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 28 Jun 2011 16:11:10 -0700
Subject: [Freeswitch-users] pause and restart while recording?
In-Reply-To: <4E0A2407.3010704@499x.com>
References: <4E0A2407.3010704@499x.com>
Message-ID: <BANLkTi=ZDqFFmbh-xbYprMMKWQU=8HKUZg@mail.gmail.com>

Dialplan script or ESL? A dp script is a bad idea for this kind of thing.
Anything that requires async ops like listening for digits while doing other
things is a candidate for ESL.

What cool new app are you working on? :)

-MC

On Tue, Jun 28, 2011 at 11:57 AM, Wes <wes-fs at 499x.com> wrote:

> I couldn't find much on this:  Is it possible in lua script to let the
> user press a key to pause recording, and press a key again to continue
> the recording, just like a traditional recorder?
>
> Thanks!
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From bryansmart at bryansmart.com  Wed Jun 29 04:47:20 2011
From: bryansmart at bryansmart.com (Bryan Smart)
Date: Tue, 28 Jun 2011 20:47:20 -0400
Subject: [Freeswitch-users] Proper prompt gain/level
In-Reply-To: <BANLkTi=+DQ2h1CeS8cqkpgZ+TCnrB7=idg@mail.gmail.com>
References: <FE081E27-8D62-4715-B7AA-64432598E013@bryansmart.com>
	<0D3AEFF37C1643B79CE3F7A7EE9EAEC3@dell9400>
	<66E81300-46E9-4C81-B2E0-123BCADDCAEB@bryansmart.com>
	<4E088A24.7010002@coppice.org>
	<3081AB39-6EF0-4C43-9AC1-7D43FCD378CD@bryansmart.com>
	<4E092ED4.60605@coppice.org>
	<2D33125A-05E0-40DE-92A2-7038940977AB@bryansmart.com>
	<4E09FA13.5030104@coppice.org>
	<88F60C8D-AFB1-46DB-9A88-4AB295A0376B@bryansmart.com>
	<BANLkTi=+DQ2h1CeS8cqkpgZ+TCnrB7=idg@mail.gmail.com>
Message-ID: <37721EF7-5FCD-43BF-A988-90C806E9E11F@bryansmart.com>

Thanks, Anthony.

I downloaded

http://files.freeswitch.org/freeswitch-sounds-en-us-callie-48000-1.0.16.tar.gz

and

http://files.freeswitch.org/freeswitch-sounds-en-us-callie-32000-1.0.16.tar.gz

I examined them with the statistical analysis tool in Sound Forge. Regardless of what DB scale is involved, both versions register in Sound Forge with identical max sample value and average RMS. This outcome doesn't seem possible if the 48000 set is sent through sox with "-v 0.2" to produce the other sets. Wouldn't the results of that processing be that the sets for other rates would be 20% as loud as the 48000 set?

The 48000 set that I discovered must not be the correct set.

By the way, the 48000 set that I downloaded also included an 8000 set. Is that expected?

Bryan

On Jun 28, 2011, at 4:42 PM, Anthony Minessale wrote:

> I don't think anyone is thinking of your post as flames.  Like I said,
> check out the original 48k files and see what the levels are vs what
> sox does to them.  I am not adverse to changing it as long as the
> majority of people agree.  Steve is very pragmatic when it comes to
> DSP.  I often make a fool of myself trying to find the right words to
> explain myself.  I don't believe he in any way is attempting to deter
> your efforts he just is a stickler for proper terminology and factual
> info.
>
>
>
> On Tue, Jun 28, 2011 at 2:26 PM, Bryan Smart <bryansmart at bryansmart.com> wrote:
>> I must be wrong about DBm0, then. I wasn't familiar with it, so I checked Wikipedia. Wikipedia isn't always right, of course.
>>
>> Wikipedia says...
>> ----------
>> dBm0 is an abbreviation for the power in dBm measured at a zero transmission level point.
>>
>> dBm0 is a concept used (amongst other areas) in audio/telephony processing since it allows a smooth integration of analog and digital chains. Notably, for
>> A-law and ?-law codecs the standards define a sequence which has a 0 dBm0 output.
>>
>> ......
>>
>> Note 2: 0 dBm0 is often replaced by or used instead of digital milliwatt or zero transmission level point.
>> ----------
>>
>> Where is my zero level transmission point? Data from a codec is not power being transmitted, only data for reproducing power at some later time. Isn't it all up to the D/A converter at the far end to determine what the power levels yielded by decoding the data will be represented relative to. If it is a hardware SIP phone, maybe the person has the volume cranked up, or turned down. Since the analog representation of the signal starts in the D/A that feeds the handset, what is considered 0? I have no idea. It sounds like this is a scale used for calibrating a D/A that feeds a pstn circuit.
>>
>> I admit feeling frustrated by this discussion. I've mixed music and mastered CDs for nearly 15 years, and I've always felt that 0DB in an entirely digital domain is a near universally understood concept. If I have a sample of 16-bit signed LPCM audio, then -32768 or 32767 represent the max amplitude that can be stored, and is what everyone that I know in pro audio calls 0DB when speaking strictly about a file, rather than a PA, in/out levels to an analog device like a tape machine, etc. If the gain of the encoded signal is boosted to a point where none of the samples are pushed beyond this range, then none of them clip. I thought that you'd think of DB in the same way, given we were talking about levels in files, but it feels like you're nit-picking or antagonizing me. It sounds like you're telling me that they can clip, even if the gain was never increased to a point where this range would be overflowed, but I have no idea how that is possible. It does not happen when I play such audio through the D/A on a computer's sound card, nor when I store it on a CD and play it back through a stereo.
>>
>> I take your point about headroom. Still, I don't feel that the current level matches caller speech. If I felt that the prompts blended well, I wouldn't have even put myself through this thread. I've connected a mix of hardware SIP phones, desktop SIP clients, and iOS SIP clients to a conference, with no modifications to the audio level of the channel, and the level of people speaking to each other in the conference is significantly louder than any prompts that are played through it. Maybe *all* of the clients are pushing audio too strongly. I first thought it was something to do with the conference, but I soon realized that the prompts were quiet everywhere, not just when played in a conference.
>>
>> I don't expect anything to change due to my personal preference, and I realize that a background in digital audio as applies to music and live recording doesn't mean that I'm not ignorant about many things that involve digital audio as it applies to telephony. I raised the issue here to see if I might be doing something wrong. If not, I wondered if the prompt levels are set based on some sort of standard by people that are wiser regarding the details than me? It seems, though, that there really isn't a standard, and the decision is someone else's personal preference. I'd rather that personal preferences never be a default. Since there isn't an official standard/guideline, I suppose that someone has to make a decision, and that decision will be influenced by their own preferences.
>>
>> I have a few suggestions to help improve Michael's sox script, but that's where I'll leave this issue. It isn't worth a big argument when I can fix it myself.
>>
>> I apologize to the list for any smoke, hints of flames, or other frustration that might have leaked through in to my posts. The sound level thing matters to me, but it is really a small thing. I enjoy Freeswitch immensely, and really appreciate everyones' efforts in producing and updating it. I've always been fascinated with phones and any type of interactive phone app, and so Asterisk, and now Freeswitch, really spark my imagination. I'm 34, and the study of phones in my early teens was my first conceptual exposure to a large network. I phreaked a bit at the time. I was fortunate (at least in one regard) to live in the US deep south, where digital switching equipment wasn't common, and so all of the old 1970's techniques weren't unavailable. That didn't last long, but it peaked my curiosity. I used VXML for a project for an employer in 2002 or so, but didn't really feel any excitement about phones, in the way that I used to, until I ran across Asterisk. Asterisk was great for the time, but I wanted to use it more for apps than a PBX, and so was quite excited to discover the different design of Freeswitch. I rarely become excited about new environments and frameworks anymore, but Freeswitch has put me back in to a fun mood of exploration and experimentation.
>>
>> Bryan
>>
>> On Jun 28, 2011, at 11:58 AM, Steve Underwood wrote:
>>
>>> On 06/28/2011 11:04 AM, Bryan Smart wrote:
>>>> I think that dBm0 only applies if we are measuring the power on an analog circuit, or at the d/a point of a digital circuit.
>>> Nope.
>>>> I was performing analysis of a digitally encoded audio file, therefore the measurement is dBFS, and 0DBFS is the point where clipping takes place. Anything below 0DBFS does not clip, even though it might seem loud to someone.
>>> What is your notion if dBFS? There are two - one where 0dBFS is a sine
>>> wave touching the limits of the number range, and the other where 0dBFS
>>> is a square wave touching those limits. They are only 3dB apart, but
>>> dbOv and dBm0 are better defined scales.
>>>
>>> Your notion is 0dBFS not clipping is based on a sine wave. For speech
>>> you need something lower because of the statistics of speech.
>>>> I stated that the peak power of the stock prompts are typically -15 to -16 (DBFS), and the short term RMS power is about -32DBFS.
>>> That statement doesn't seem to make much sense.
>>>> I don't know how to properly evaluate how DBFS will convert to DBm0. I gather that the codec and the d/a converter attenuate the level to some extent, but I don't know how much. If I have a digital file with a 1Khz test tone at -10DBFS, and play that over the pstn, what do I get out in DBm0?
>>> A sine wave touching the limits of the digital range is about +3.14dBm0.
>>>> Speaking just in terms of DBFS, -16DB is only about 15% of the potential power available before the audio reaches 0DB, and clips. Perhaps when -16DBFS is put out over the pstn, it is much closer to clipping than it would be in an entirely digital domain. I don't know enough to make that determination. Do you know?
>>> Yes, and you should know, since I told you last time.
>>>> I'm not for clipping and distortion. However, too little power can cause another problem: limited dynamic range and increased dithering artifacts. If audio is quiet on a pstn phone, then the person with the phone might be able to increase the level by turning up the phone's volume, if it has one. However, that raises the noise floor. Companding might make 8-bit channels sound a bit like 14-bit or 15-bit channels in terms of a low noise floor and decreased dithering artifacts, but it is still just an 8-bit channel. Companding hides most of the dithering artifacts in strong signals, but it magnifies them in weak signals. That's why G711 is fairly clear, but will sound scratchy if you put faint signals in to it and try to amplify them back up to normal levels. Over G711 or any pstn call, as you decrease the level of the audio going across, the scratchy dithering static obscures an increasing amount of the audio.
>>> That paragraph is utter drivel. The companding of G.711 ensures that the
>>> audio quality is roughly maintained from clipping down to -45dBm0 or so.
>>> Below that the quality falls, just like most other digital coding scheme
>>> running out of bits. The FS prompts aren't nearly that quiet.
>>>> This gets worse if you compound it by stacking codecs. Consider if someone calls your IVR from a cell phone, and you play quiet prompts to them. The audio is first passed through G711, where the low gain means that scratchy dithering artifacts are added. Then, it is encoded to GSM for the cell phone,. GSM uses linear predictive coding, and, being tuned for voice, it is not optimized for having smooth waveforms interrupted periodically with random excursions. Thats part of why cell phone calls sound so bad if there is lots of background noise.
>>> Again, this is drivel.
>>>> Anyway, clipping is to be avoided, but simply reducing levels dramatically creates other quality problems on a channel that uses companding. You trade distorted audio for scratchy audio. Same thing happened with cassette tapes that used Dolby noise reduction.
>>> You really are just making this up as you go along, aren't you.
>>>> Maybe you can help clear up my understanding of how DBFS in a digital file will work out in DBm0 on a pstn line. As things stand, though, I think we have lots of room to increase the level of the prompts before we reach a point where clipping is an issue.
>>> I tried playing with a few of the prompts. The peaks of the words are
>>> about -20dBm0, or -26dBOv. That's probably -23 or -26 on your scale.
>>> Every word peaks to a similar level, as prompts are spoken in a pretty
>>> flat voice. They seem to be about 15dB away clipping, which would mean
>>> the crest factor is about 11dB for the Callie voice, which seems a
>>> little low for speech. You usually need to ride a little more than 13dB
>>> above the short term power of most voices to bring the crossing rate
>>> close to zero.
>>>
>>> So, there is about 15dB of headroom which could be used to increase the
>>> speech level without clipping. Whether doing that is a good idea is
>>> questionable. The prompts are at about normal speech level. In normal
>>> speech the codecs have plenty of headroom to cope with people getting
>>> agitated and shouting, producing only modest amounts of clipping. Is
>>> cranking all the prompts up to shouting level appropriate?
>>>
>>> Steve
>>>
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
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> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From bryansmart at bryansmart.com  Wed Jun 29 06:12:43 2011
From: bryansmart at bryansmart.com (Bryan Smart)
Date: Tue, 28 Jun 2011 22:12:43 -0400
Subject: [Freeswitch-users] Proper prompt gain/level
In-Reply-To: <BANLkTi=9-B3R74voiHLV6RgcvvkqwS6aZQ@mail.gmail.com>
References: <FE081E27-8D62-4715-B7AA-64432598E013@bryansmart.com>
	<0D3AEFF37C1643B79CE3F7A7EE9EAEC3@dell9400>
	<66E81300-46E9-4C81-B2E0-123BCADDCAEB@bryansmart.com>
	<4E088A24.7010002@coppice.org>
	<3081AB39-6EF0-4C43-9AC1-7D43FCD378CD@bryansmart.com>
	<BANLkTikTK3JW=jDFNqkZqPxhic_bzL=uCQ@mail.gmail.com>
	<BANLkTimSuFpBXwANu8qc2q24z-xNYdcJnw@mail.gmail.com>
	<1BB0CF88-DA61-4AB6-9CC6-328CCB9846AD@bryansmart.com>
	<BANLkTi=9-B3R74voiHLV6RgcvvkqwS6aZQ@mail.gmail.com>
Message-ID: <7D766BBE-F3D1-4F9C-B7B9-A44B16EB2BF6@bryansmart.com>

Hi, Michael.

Thanks for posting this.

Have you considered using the higher quality sample rate conversion features in new versions of sox?

A starting place for info is here:
http://sox.sourceforge.net/SoX/Resampling

Perhaps change:

sox -v 0.2 $base_dir/$dir/$filename -r $rate -c 1 $tmp_dir/$voice_dir/$dir/$rate/$filename

to:

sox $base_dir/$dir/$filename -c 1 $tmp_dir/$voice_dir/$dir/$rate/$filename rate -v -I $rate dither

rate -v -I $rate: Replaces -r $rate. Uses the newer "very high quality" mode (instead of "high quality") with Intermediate Phase response (instead of linear). This setting is superior to the default, but slower to process.

dither: Applies basic triangular dithering. There are other dithering strategies, but, based on what I've read and experienced, they aren't as useful when down-sampling to low rates.

Might produce slightly higher quality files, the next time it is necessary to regenerate them.

Bryan

On Jun 28, 2011, at 11:18 AM, Michael Collins wrote:

You can download the 48kHz files from files.freeswitch.org<http://files.freeswitch.org/>. The script itself is nothing special - all it does is cycle through the target sampling rates and run each file through sox. Here's a copy of the script:

base_dir="48000"
rates="48000 32000 16000 8000"
version="1.0.16"
voice="en-us-callie"
voice_dir="en/us/callie"
tar_path="../../.."
tmp_dir="tmp"
CWD=`pwd`

for rate in $rates; do \
  mkdir -p $tmp_dir/$voice_dir
  for dir in `ls $base_dir`; do \
    test -d $tmp_dir/$voice_dir/$dir/$rate || mkdir -p $tmp_dir/$voice_dir/$dir/$rate; \
    for filename in `ls $base_dir/$dir`; do \
      echo sox -v 0.2 $base_dir/$dir/$filename -r $rate -c 1 $tmp_dir/$voice_dir/$dir/$rate/$filename; \
      sox -v 0.2 $base_dir/$dir/$filename -r $rate -c 1 $tmp_dir/$voice_dir/$dir/$rate/$filename; \
    done ; \
  done ; \
  cd $tmp_dir
  echo tar -cvzf $CWD/$tar_path/freeswitch-sounds-$voice-$rate-$version.tar.gz *; \
  tar -cvzf $CWD/$tar_path/freeswitch-sounds-$voice-$rate-$version.tar.gz *; \
  cd -
  rm -rf $tmp_dir
done

cd $tar_path
for rate in $rates; do \
  openssl dgst -sha1 freeswitch-sounds-$voice-$rate-$version.tar.gz > freeswitch-sounds-$voice-$rate-$version.tar.gz.sha1 ; \
  openssl dgst -md5 freeswitch-sounds-$voice-$rate-$version.tar.gz > freeswitch-sounds-$voice-$rate-$version.tar.gz.md5 ; \
done
cd $CWD

Let me know if you have any suggestions.
-MC

On Mon, Jun 27, 2011 at 8:25 PM, Bryan Smart <bryansmart at bryansmart.com<mailto:bryansmart at bryansmart.com>> wrote:
-v is a multiplier. So, -v 0.2 reduces gain to 20% of original.

It is typical to supply professionally recorded audio with peaks boosted to 0DBFS. That is probably how the prompts were originally supplied. It makes since that -v 0.2 would reduce the max power to 20%, around -15 to -16 DBFS.

The FS build process downloads pre-processed versions of these files. Is there somewhere online that I can find the source recordings+script that produces the pre-processed files that are downloaded by the build process?

Bryan

On Jun 27, 2011, at 7:32 PM, Michael Collins wrote:



On Mon, Jun 27, 2011 at 3:07 PM, Anthony Minessale <anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>> wrote:
Have a look at the 48khz versions of the files, those should be the originals.
Someone runs a batch sox command on them to get them to the other levels.

Just for the record, the sox command used to normalize the sounds is:
sox -v 0.2 $file_in -r $rate -c 1 $file_out

The -v format is the "volume" format and if I read the sox man page correctly it means a "linear amplitude adjustment". What I can't tell from the man page is whether .2 means "reduce by 20%" or "reduce to 20%" or something else. In any case, like Tony says, we are not picky about this as long as it's not pointlessly loud.

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

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


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

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


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

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

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From sharad at coraltele.com  Wed Jun 29 11:47:25 2011
From: sharad at coraltele.com (sharad)
Date: Wed, 29 Jun 2011 13:17:25 +0530
Subject: [Freeswitch-users] SIP Profile
References: <mailman.11637.1309313596.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <480E4906DC4F48E2AA9B823F5ADBCD29@sharad>

Hi,

I want my FS to entertain only those calls which are coming from a specific IP address . 

For this, I have defined a profile in vars.xml as 

        <X-PRE-PROCESS cmd="set" data="profile_00=192.168.4.10"/>.

        & another line in public.xml as 

        <extension name="transfer_to_context_01">

        <condition field="${sip_from_host}" expression="$${profile_00}">

        <action application="transfer" data="${destination_number} XML sharad"/>

        </condition>

        </extension>

As per my understanding, if call is coming from 192.168.4.10, that call should be processed as per the context `sharad'.....correct ?

Yes, it works like this. Upto here all is ok.

But if the call is coming from 192.168.100 - 192.168.4.109, these calls also are processed as per context `sharad'. 

Means entire IP is not matched by condition field. 

Is there any solution for this. 

Requesting all of you to help me in this regard.

Regards

Sharad
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From peter.olsson at visionutveckling.se  Wed Jun 29 12:03:59 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 29 Jun 2011 10:03:59 +0200
Subject: [Freeswitch-users] SIP Profile
In-Reply-To: <480E4906DC4F48E2AA9B823F5ADBCD29@sharad>
References: <mailman.11637.1309313596.24555.freeswitch-users@lists.freeswitch.org>
	<480E4906DC4F48E2AA9B823F5ADBCD29@sharad>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59E54FA23A@cooper>

You need to escape the dots, ie 192\.168\.4\.10. That should make things work as expected.

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r sharad
Skickat: den 29 juni 2011 09:47
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] SIP Profile


Hi,

I want my FS to entertain only those calls which are coming from a specific IP address .

For this, I have defined a profile in vars.xml as

        <X-PRE-PROCESS cmd="set" data="profile_00=192.168.4.10"/>.

        & another line in public.xml as

        <extension name="transfer_to_context_01">

        <condition field="${sip_from_host}" expression="$${profile_00}">

        <action application="transfer" data="${destination_number} XML sharad"/>

        </condition>

        </extension>

As per my understanding, if call is coming from 192.168.4.10, that call should be processed as per the context `sharad'.....correct ?

Yes, it works like this. Upto here all is ok.

But if the call is coming from 192.168.100 - 192.168.4.109, these calls also are processed as per context `sharad'.

Means entire IP is not matched by condition field.

Is there any solution for this.

Requesting all of you to help me in this regard.

Regards

Sharad
!DSPAM:4e0ada5632766217057301!
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From steveayre at gmail.com  Wed Jun 29 12:47:01 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 29 Jun 2011 09:47:01 +0100
Subject: [Freeswitch-users] SIP Profile
In-Reply-To: <480E4906DC4F48E2AA9B823F5ADBCD29@sharad>
References: <mailman.11637.1309313596.24555.freeswitch-users@lists.freeswitch.org>
	<480E4906DC4F48E2AA9B823F5ADBCD29@sharad>
Message-ID: <BANLkTi=UUZRXcQACEtJMpZDkrUL4_U1-BQ@mail.gmail.com>

There are also a few other ways to handle this before the dialplan:

1) ACL on the profile that restricts dialing in to 192.168.4.10
2) User account with cidr="192.168.4.10" and <variable name="user_context"
value="sharad"/>

First would prevent anyone calling in except from your IP.
Second would automatically authenticate anything from that IP as your user
and handle it in the shared context, anyone else would be calling in
unauthenticated into the default context.

-Steve



On 29 June 2011 08:47, sharad <sharad at coraltele.com> wrote:

> **
>
> Hi,
>
> I want my FS to entertain only those calls which are coming from a specific
> IP address .
>
> For this, I have defined a profile in vars.xml as
>
>         <X-PRE-PROCESS cmd="set" data="profile_00=192.168.4.10"/>.
>
>         & another line in public.xml as
>
>         <extension name="transfer_to_context_01">
>
>         <condition field="${sip_from_host}" expression="$${profile_00}">
>
>         <action application="transfer" data="${destination_number} XML
> sharad"/>
>
>         </condition>
>
>         </extension>
>
> As per my understanding, if call is coming from 192.168.4.10, that call
> should be processed as per the context `sharad'.....correct ?
>
> Yes, it works like this. Upto here all is ok.
>
> But if the call is coming from 192.168.100 - 192.168.4.109, these calls
> also are processed as per context `sharad'.
>
> Means entire IP is not matched by condition field.
>
> Is there any solution for this.
>
> Requesting all of you to help me in this regard.
>
> Regards
>
> Sharad
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mays.david at gmail.com  Wed Jun 29 13:35:07 2011
From: mays.david at gmail.com (David Ma)
Date: Wed, 29 Jun 2011 17:35:07 +0800
Subject: [Freeswitch-users] Leg-A is automatically disconnected on Leg-B
 orginate failure
In-Reply-To: <BANLkTi=ZvOZ99qVxUeJAtpjtwP5r=hfVkg@mail.gmail.com>
References: <1308213151543-6482192.post@n2.nabble.com>
	<BANLkTi=xYgBTPVGxVGmk9f2=bmX5X2vF=Q@mail.gmail.com>
	<BANLkTikhPmjN5HeeJHBnQt1ivA8zbtCnYA@mail.gmail.com>
	<BANLkTi=m7oMix1ZRC9URtRuAbsG2fWyhMQ@mail.gmail.com>
	<BANLkTimxBU8m311LVKMWcFOjukwLg+ZUGQ@mail.gmail.com>
	<BANLkTinZJf3BxB7QHefnmOvxQmVW2r4dTg@mail.gmail.com>
	<BANLkTimiExLew_1cjMuQVpF3CKkR8mrGXQ@mail.gmail.com>
	<BANLkTi=ZvOZ99qVxUeJAtpjtwP5r=hfVkg@mail.gmail.com>
Message-ID: <BANLkTimj4iSWFiHuBSKskz7NjF2bCEWnaw@mail.gmail.com>

Hi Anthony,

Thanks very much for the information. I appreciate your advice. It is great
to learn about such a parameter.

FS wiki has only a little description about this parameter. Is there any
side effect, or caution for using such a parameter?

Thanks,
D.Ma

On Wed, Jun 29, 2011 at 12:08 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> The cases where you have this problem arise because you are
> originating to the B leg who answers 183 (early media) then hangs up
> before the call was answered.
>
> set the channel variable uuid_bridge_continue_on_cancel=true on the A
> leg to change the behavior to what you want.
>
>
> On Mon, Jun 27, 2011 at 10:09 AM, Michael Collins <msc at freeswitch.org>
> wrote:
> > This makes my eyes bleed. Can you please put this on
> > pastebin.freeswitch.org? Use "FreeSWITCH Log" as the syntax highlight.
> > -MC
> >
> > On Sun, Jun 26, 2011 at 11:27 PM, David Ma <mays.david at gmail.com> wrote:
> >>
> >> Hi Michael,
> >>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ankitwalia4u at gmail.com  Wed Jun 29 16:05:01 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Wed, 29 Jun 2011 17:35:01 +0530
Subject: [Freeswitch-users] mod_sndfile.c - path separator for windows -
	system error
Message-ID: <BANLkTinvoLS7WYj5Amf_y9cNsyOm2gXmDQ@mail.gmail.com>

While executing a lua script, I got this error. I am running FS on window 7.

2011-06-29 17:27:52.265291 [ERR] mod_sndfile.c:195 Error Opening File
[C:\ankitw
alia\FreeSwitch\freeswitch-1.0.6\Debug\sounds/en/us/callie\blah.wav] [System
err
or : The system cannot find the file specified.
]

This seems to be because of path seperator used as "/" instead of "\" for
windows in mod_sndfile.c file.

Thanks
Ankit

-- 
Life is like a rose its upto u feel it as its fragrance or thorns
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From steveayre at gmail.com  Wed Jun 29 16:12:24 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 29 Jun 2011 13:12:24 +0100
Subject: [Freeswitch-users] mod_sndfile.c - path separator for windows -
 system error
In-Reply-To: <BANLkTinvoLS7WYj5Amf_y9cNsyOm2gXmDQ@mail.gmail.com>
References: <BANLkTinvoLS7WYj5Amf_y9cNsyOm2gXmDQ@mail.gmail.com>
Message-ID: <BANLkTindaukmFPhPXxZpr+nfUfpq6kOzzUjRKjfD9DXK9+LYUQ@mail.gmail.com>

Shouldn't be - all recent versions of Windows accept / as a path separator.

Are you sure the file exists in that path?

-Steve


On 29 June 2011 13:05, ankIT WALiA <ankitwalia4u at gmail.com> wrote:

> While executing a lua script, I got this error. I am running FS on window
> 7.
>
> 2011-06-29 17:27:52.265291 [ERR] mod_sndfile.c:195 Error Opening File
> [C:\ankitw
> alia\FreeSwitch\freeswitch-1.0.6\Debug\sounds/en/us/callie\blah.wav]
> [System err
> or : The system cannot find the file specified.
> ]
>
> This seems to be because of path seperator used as "/" instead of "\" for
> windows in mod_sndfile.c file.
>
> Thanks
> Ankit
>
> --
> Life is like a rose its upto u feel it as its fragrance or thorns
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From peter.olsson at visionutveckling.se  Wed Jun 29 16:20:10 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 29 Jun 2011 14:20:10 +0200
Subject: [Freeswitch-users] mod_sndfile.c - path separator for windows
	-	system error
In-Reply-To: <BANLkTinvoLS7WYj5Amf_y9cNsyOm2gXmDQ@mail.gmail.com>
References: <BANLkTinvoLS7WYj5Amf_y9cNsyOm2gXmDQ@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59E54FA402@cooper>

My guess is that the file doesn't exist, \ or / doesn't matter in Windows anymore.

However, you might also change the sound_prefix in vars.xml, if you really want to change the path to something else.

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r ankIT WALiA
Skickat: den 29 juni 2011 14:05
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] mod_sndfile.c - path separator for windows - system error

While executing a lua script, I got this error. I am running FS on window 7.

2011-06-29 17:27:52.265291 [ERR] mod_sndfile.c:195 Error Opening File [C:\ankitw
alia\FreeSwitch\freeswitch-1.0.6\Debug\sounds/en/us/callie\blah.wav] [System err
or : The system cannot find the file specified.
]

This seems to be because of path seperator used as "/" instead of "\" for windows in mod_sndfile.c file.

Thanks
Ankit

--
Life is like a rose its upto u feel it as its fragrance or thorns
!DSPAM:4e0b166332769092317685!
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From ankitwalia4u at gmail.com  Wed Jun 29 16:35:06 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Wed, 29 Jun 2011 18:05:06 +0530
Subject: [Freeswitch-users] mod_sndfile.c - path separator for windows -
 system error
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59E54FA402@cooper>
References: <BANLkTinvoLS7WYj5Amf_y9cNsyOm2gXmDQ@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E54FA402@cooper>
Message-ID: <BANLkTi=0giRgkQkqEqKO3aarObNpxnAF7g@mail.gmail.com>

I'm sorry. I did place the file there. My mistake.

On Wed, Jun 29, 2011 at 5:50 PM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> My guess is that the file doesn?t exist, \ or / doesn?t matter in Windows
> anymore.****
>
> ** **
>
> However, you might also change the sound_prefix in vars.xml, if you really
> want to change the path to something else.****
>
> ** **
>
> /Peter****
>
> ** **
>
> ** **
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *ankIT WALiA
> *Skickat:* den 29 juni 2011 14:05
> *Till:* FreeSWITCH Users Help
> *?mne:* [Freeswitch-users] mod_sndfile.c - path separator for windows -
> system error****
>
> ** **
>
> While executing a lua script, I got this error. I am running FS on window
> 7.
>
> 2011-06-29 17:27:52.265291 [ERR] mod_sndfile.c:195 Error Opening File
> [C:\ankitw
> alia\FreeSwitch\freeswitch-1.0.6\Debug\sounds/en/us/callie\blah.wav]
> [System err
> or : The system cannot find the file specified.
> ]
>
> This seems to be because of path seperator used as "/" instead of "\" for
> windows in mod_sndfile.c file.
>
> Thanks
> Ankit
>
> --
> Life is like a rose its upto u feel it as its fragrance or thorns
> !DSPAM:4e0b166332769092317685! ****
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Life is like a rose its upto u feel it as its fragrance or thorns
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From avi at avimarcus.net  Wed Jun 29 16:41:46 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 29 Jun 2011 15:41:46 +0300
Subject: [Freeswitch-users] SIP Profile
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59E54FA23A@cooper>
References: <mailman.11637.1309313596.24555.freeswitch-users@lists.freeswitch.org>
	<480E4906DC4F48E2AA9B823F5ADBCD29@sharad>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E54FA23A@cooper>
Message-ID: <BANLkTimwO7jxfm1fpsNaoVaPEnohbmdYsg@mail.gmail.com>

Yes, if you take this route, do what Peter said.. but mostly the issue is:

?<X-PRE-PROCESS cmd="set" data="profile_00=192.168.4.10"/>.
<condition field="${sip_from_host}" expression="$${profile_00}">

So the match is: "it must CONTAIN the variable" - what you want is "it
must BE the variable"

To do it properly would be: <condition field="${sip_from_host}"
expression="^$${profile_00}$">

Anyway, Steven's solution is much better.
-Avi Marcus

On Wed, Jun 29, 2011 at 11:03 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
>
> You need to escape the dots, ie 192\.168\.4\.10. That should make things work as expected.
>
>
>
> /Peter
>
>
>
>
>
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r sharad
> Skickat: den 29 juni 2011 09:47
> Till: freeswitch-users at lists.freeswitch.org
> ?mne: [Freeswitch-users] SIP Profile
>
>
>
> Hi,
>
> I want my FS to entertain only those calls which are coming from a specific IP address .
>
> For this,?I have defined a profile in vars.xml as
>
> ??? ??? <X-PRE-PROCESS cmd="set" data="profile_00=192.168.4.10"/>.
>
> ??? ??? & another line in public.xml as
>
> ??? ??? <extension name="transfer_to_context_01">
>
> ??? ??? <condition field="${sip_from_host}" expression="$${profile_00}">
>
> ??? ??? <action application="transfer" data="${destination_number} XML sharad"/>
>
> ??? ??? </condition>
>
> ??? ??? </extension>
>
> As per my understanding, if call is coming from 192.168.4.10, that call should be processed as per the context `sharad'.....correct ?
>
> Yes, it works like this. Upto here all is ok.
>
> But if the call is coming from 192.168.100 - 192.168.4.109, these calls also are processed as per context `sharad'.
>
> Means entire IP is not matched by condition field.
>
> Is there any solution for this.
>
> Requesting all of you to help me in this regard.
>
> Regards
>
> Sharad
>
> !DSPAM:4e0ada5632766217057301!
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From abid_freeswitch at live.com  Wed Jun 29 17:59:41 2011
From: abid_freeswitch at live.com (Abid Saleem)
Date: Wed, 29 Jun 2011 19:59:41 +0600
Subject: [Freeswitch-users] Load Balance Trunks
Message-ID: <col109-w22D1131E50A2495C1521C9F4590@phx.gbl>


Hi All,
Please help me the following 2 issues.

1- I have 100 Trunks from my SIP Provider. My provider restricts me to send only 120 minutes call duration per Trunk per each day. This means around 30 calls per Trunk per day with an ACD of 3 minutes. Please help me how to configure this if it is possible?
2- I need to send P-Preferred-Identity<sip:xxxxxxxxx at domain.com> in SIP header for each Trunk while dialing out to provider. Please help how to configure this.
Thanks in Advance.
Rgrds-----------Abid SaleemTechnical Manager NGNTerminus Technologies 		 	   		  
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From wes-fs at 499x.com  Wed Jun 29 18:00:00 2011
From: wes-fs at 499x.com (Wes)
Date: Wed, 29 Jun 2011 09:00:00 -0500
Subject: [Freeswitch-users] pause and restart while recording?
In-Reply-To: <BANLkTi=ZDqFFmbh-xbYprMMKWQU=8HKUZg@mail.gmail.com>
References: <4E0A2407.3010704@499x.com>
	<BANLkTi=ZDqFFmbh-xbYprMMKWQU=8HKUZg@mail.gmail.com>
Message-ID: <4E0B2FE0.1070801@499x.com>

I'm writing the script in LUA... not sure how to answer your question RE 
dialplan or ESL...

The lua script is triggered via the socket library from a php page, and 
the arguments to the lua script are things like the phone number to 
dial, and an id number to track things back.

The application is pretty simple, it just needs to call the desired 
number, play instructions via voice commands, take a recording, allow 
user to review the recording, rerecord it, and then finally submit it.  
It's going to be for medical dictation.  It's pretty much done except 
for the pause feature.

Searching around the archives for the "pause during recording" 
functionality, I found a few people discussing the need for pause in 
relation to a transcription app, but couldn't figure out what came of it.

I'm pretty close, as I already have a loop that will stop the recording 
on a "#" keypress, so if it could just pause recording on a different 
keypress, that would be great...

Thanks for the help!

On 6/28/2011 6:11 PM, Michael Collins wrote:
> Dialplan script or ESL? A dp script is a bad idea for this kind of 
> thing. Anything that requires async ops like listening for digits 
> while doing other things is a candidate for ESL.
>
> What cool new app are you working on? :)
>
> -MC
>
> On Tue, Jun 28, 2011 at 11:57 AM, Wes <wes-fs at 499x.com 
> <mailto:wes-fs at 499x.com>> wrote:
>
>     I couldn't find much on this:  Is it possible in lua script to let the
>     user press a key to pause recording, and press a key again to continue
>     the recording, just like a traditional recorder?
>
>     Thanks!
>
>     _______________________________________________
>     Join us at ClueCon 2011, Aug 9-11, Chicago
>     http://www.cluecon.com 877-7-4ACLUE
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From avi at avimarcus.net  Wed Jun 29 18:26:38 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 29 Jun 2011 17:26:38 +0300
Subject: [Freeswitch-users] Load Balance Trunks
In-Reply-To: <col109-w22D1131E50A2495C1521C9F4590@phx.gbl>
References: <col109-w22D1131E50A2495C1521C9F4590@phx.gbl>
Message-ID: <BANLkTi=eN=S3+DYjC1Vozd0yaC2Jxkbfcw@mail.gmail.com>

Hi Abid.. see inline.

On Wed, Jun 29, 2011 at 4:59 PM, Abid Saleem <abid_freeswitch at live.com>
wrote:
> Hi All,
> Please help me the following 2 issues.
>
> 1- I have 100 Trunks from my SIP Provider. My provider restricts me to
send
> only 120 minutes call duration per Trunk per each day. This means around
30
> calls per Trunk per day with an ACD of 3 minutes. Please help me how to
> configure this if it is possible?

You can use: http://wiki.freeswitch.org/wiki/Mod_distributor

OR, a more precise way.. scrape your CDRs and then query them before making
calls.
Or, have a lua script that picks a random one from the list that a cron
updates every 30 minutes to use slightly less resources.

> 2- I need to send P-Preferred-Identity<sip:xxxxxxxxx at domain.com> in SIP
> header for each Trunk while dialing out to provider. Please help how to
> configure this.
http://wiki.freeswitch.org/wiki/Variable_sip_cid_type has some info on using
PID.
You can use "sofia profile external siptrace on" to see your SIP messages
from FS to debug.

-Avi
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From jonas at jonasborjesson.com  Wed Jun 29 18:40:46 2011
From: jonas at jonasborjesson.com (Jonas Borjesson)
Date: Wed, 29 Jun 2011 07:40:46 -0700
Subject: [Freeswitch-users] Copy custom SIP headers -
 {sip_copy_custom_headers=true} doesn't seem to be working
In-Reply-To: <BANLkTimR7EAhpxaX-hRe128tq4VEF=yoeQ@mail.gmail.com>
References: <BANLkTikNA_oA_yswO+7KZ5ir7YUVDNjM2A@mail.gmail.com>
	<BANLkTimR7EAhpxaX-hRe128tq4VEF=yoeQ@mail.gmail.com>
Message-ID: <BANLkTikQBuGVLzb+e4tOSnqo41NJQjzdag@mail.gmail.com>

Hi,

Yep, downloaded and verified that it works on latest and greatest. Thanks!

/Jonas

On Tue, Jun 28, 2011 at 8:57 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Problem seems to be we did not support final responses only 1xx and
> 200. I think I missed that part of the other guy's report.
> Update again to lastest GIT and it should work.
>
> This is one reason we need JIRA for issues of this type because it's
> easy to lose track of email threads in a big list like this.
>
>
>
> On Tue, Jun 28, 2011 at 2:19 AM, Jonas Borjesson
> <jonas at jonasborjesson.com> wrote:
>> Hi all,
>>
>> As many others, I need to copy customer X-headers from leg B back over
>> to leg A and have found a few threads talking about using the
>> sip_copy_custom_headers=true with the dial string but this does not
>> seem to be working. I am doing exactly what is described in the thread
>> below but like that guy, it does not work for me either and that
>> thread never came to any conclusion. Does anyone know the status of
>> this feature or what I may be doing wrong? Any help is greatly
>> appreciated!
>>
>> Of course, I tried on the latest and greatest FS -
>> 1.0.head-git-4962542 2011-06-27 10-15-03 -0500
>>
>> The tread talking about this very feature:
>> http://comments.gmane.org/gmane.comp.telephony.freeswitch.user/36287
>>
>> And documentation:
>> http://wiki.freeswitch.org/wiki/Variable_sip_copy_custom_headers
>>
>> Thanks,
>>
>> /Jonas
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From anthony.minessale at gmail.com  Wed Jun 29 18:53:46 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 29 Jun 2011 09:53:46 -0500
Subject: [Freeswitch-users] Leg-A is automatically disconnected on Leg-B
 orginate failure
In-Reply-To: <BANLkTimj4iSWFiHuBSKskz7NjF2bCEWnaw@mail.gmail.com>
References: <1308213151543-6482192.post@n2.nabble.com>
	<BANLkTi=xYgBTPVGxVGmk9f2=bmX5X2vF=Q@mail.gmail.com>
	<BANLkTikhPmjN5HeeJHBnQt1ivA8zbtCnYA@mail.gmail.com>
	<BANLkTi=m7oMix1ZRC9URtRuAbsG2fWyhMQ@mail.gmail.com>
	<BANLkTimxBU8m311LVKMWcFOjukwLg+ZUGQ@mail.gmail.com>
	<BANLkTinZJf3BxB7QHefnmOvxQmVW2r4dTg@mail.gmail.com>
	<BANLkTimiExLew_1cjMuQVpF3CKkR8mrGXQ@mail.gmail.com>
	<BANLkTi=ZvOZ99qVxUeJAtpjtwP5r=hfVkg@mail.gmail.com>
	<BANLkTimj4iSWFiHuBSKskz7NjF2bCEWnaw@mail.gmail.com>
Message-ID: <BANLkTikyyAZaJzQx2tbiVOxHxx5mCbUdvQ@mail.gmail.com>

The only side effect is that it will give you the behavior you want in
this particular situation.


On Wed, Jun 29, 2011 at 4:35 AM, David Ma <mays.david at gmail.com> wrote:
> Hi Anthony,
>
> Thanks very much for the information. I appreciate your advice. It is great
> to learn about such a parameter.
>
> FS wiki has only a little description about this parameter. Is there any
> side effect, or caution for using such a parameter?
>
> Thanks,
> D.Ma
>
> On Wed, Jun 29, 2011 at 12:08 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> The cases where you have this problem arise because you are
>> originating to the B leg who answers 183 (early media) then hangs up
>> before the call was answered.
>>
>> set the channel variable uuid_bridge_continue_on_cancel=true on the A
>> leg to change the behavior to what you want.
>>
>>
>> On Mon, Jun 27, 2011 at 10:09 AM, Michael Collins <msc at freeswitch.org>
>> wrote:
>> > This makes my eyes bleed. Can you please put this on
>> > pastebin.freeswitch.org? Use "FreeSWITCH Log" as the syntax highlight.
>> > -MC
>> >
>> > On Sun, Jun 26, 2011 at 11:27 PM, David Ma <mays.david at gmail.com> wrote:
>> >>
>> >> Hi Michael,
>> >>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From david.ponzone at ipeva.fr  Wed Jun 29 19:32:38 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 29 Jun 2011 17:32:38 +0200
Subject: [Freeswitch-users] pause and restart while recording?
In-Reply-To: <4E0B2FE0.1070801@499x.com>
References: <4E0A2407.3010704@499x.com>
	<BANLkTi=ZDqFFmbh-xbYprMMKWQU=8HKUZg@mail.gmail.com>
	<4E0B2FE0.1070801@499x.com>
Message-ID: <3DE0409A-DDEE-4D09-9C55-F184902EA64A@ipeva.fr>

That's interesting, but what does FS do if you record over an existing file ? Does it append or does it overwrite it ?

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

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

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




Le 29/06/2011 ? 16:00, Wes a ?crit :

> I'm writing the script in LUA... not sure how to answer your question RE dialplan or ESL...
> 
> The lua script is triggered via the socket library from a php page, and the arguments to the lua script are things like the phone number to dial, and an id number to track things back.
> 
> The application is pretty simple, it just needs to call the desired number, play instructions via voice commands, take a recording, allow user to review the recording, rerecord it, and then finally submit it.  It's going to be for medical dictation.  It's pretty much done except for the pause feature.
> 
> Searching around the archives for the "pause during recording" functionality, I found a few people discussing the need for pause in relation to a transcription app, but couldn't figure out what came of it.
> 
> I'm pretty close, as I already have a loop that will stop the recording on a "#" keypress, so if it could just pause recording on a different keypress, that would be great... 
> 
> Thanks for the help!
> 
> On 6/28/2011 6:11 PM, Michael Collins wrote:
>> 
>> Dialplan script or ESL? A dp script is a bad idea for this kind of thing. Anything that requires async ops like listening for digits while doing other things is a candidate for ESL. 
>> 
>> What cool new app are you working on? :)
>> 
>> -MC
>> 
>> On Tue, Jun 28, 2011 at 11:57 AM, Wes <wes-fs at 499x.com> wrote:
>> I couldn't find much on this:  Is it possible in lua script to let the
>> user press a key to pause recording, and press a key again to continue
>> the recording, just like a traditional recorder?
>> 
>> Thanks!
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at ipeva.fr  Wed Jun 29 19:41:41 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 29 Jun 2011 17:41:41 +0200
Subject: [Freeswitch-users] Load Balance Trunks
In-Reply-To: <col109-w22D1131E50A2495C1521C9F4590@phx.gbl>
References: <col109-w22D1131E50A2495C1521C9F4590@phx.gbl>
Message-ID: <C71A4485-E533-48AE-AC3F-C59D7A1E79CA@ipeva.fr>

> 1- I have 100 Trunks from my SIP Provider. My provider restricts me to send only 120 minutes call duration per Trunk per each day. This means around 30 calls per Trunk per day with an ACD of 3 minutes. Please help me how to configure this if it is possible?

Can't you find another provider with no such ... limitations ?

You can do what Avi told you, or you can write your own piece of LUA script which would check in a DB which trunk to use, and then would bridge using this trunk, and will update the DB after hangup to add the call duration of the last call to the total amount for trunk X and current day (that would have to be done with the hangup_hook thingy, probably).

> 2- I need to send P-Preferred-Identity<sip:xxxxxxxxx at domain.com> in SIP header for each Trunk while dialing out to provider. Please help how to configure this.
> 
bridge {sip_cid_type=none,sip_h_P-Preferred-Identity=XXXXXXX}sofia/.....  -> I am sure of this one
or
bridge {sip_cid_type=pid}sofia/.... -> never tested

> Thanks in Advance.
> 
> Rgrds
> -----------
> Abid Saleem
> Technical Manager NGN
> Terminus Technologies
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From lists at ticm.com  Wed Jun 29 13:49:22 2011
From: lists at ticm.com (Bret-lists Watson)
Date: Wed, 29 Jun 2011 17:49:22 +0800
Subject: [Freeswitch-users] halp! newb question on setting up providers..
Message-ID: <BANLkTikUAUUkR0FZ=pu+9GEN9=zVhAFkaQ@mail.gmail.com>

Hi All,
getting freeswitch up and running - heaps easier than asterisk.... but..

I can't seem to work out what and where to put things for configuring my
provider.. there seems to be a number of ways to do it..

currently I have it set in my vars.xml under default_provider..

 <X-PRE-PROCESS cmd="set" data="default_provider=sip.pennytel.com"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_username=6189xxxxxx"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_password=xxxxxx"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_from_domain=
sip.pennytel.com"/>
  <!-- true or false -->
  <X-PRE-PROCESS cmd="set" data="default_provider_register=true"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_contact=sofia/internal/
sip:102 at 192.168.1.22:5060"/>

when I try and ring my mobile I get..

2011-06-29 17:39:42.269994 [WARNING] sofia_reg.c:1329 SIP auth challenge
(INVITE) on sofia profile 'internal' for [041xxxxxxx at 192.168.1.76] from ip
192.168.1.22
2011-06-29 17:39:42.610032 [NOTICE] switch_channel.c:890 New Channel
sofia/internal/102 at 192.168.1.76 [cda831fb-e8f5-4c64-936a-1562691f15cf]
2011-06-29 17:39:42.657134 [INFO] mod_dialplan_xml.c:336 Processing Kitchen
<102>->041xxxxxxx in context default
2011-06-29 17:39:42.745992 [ERR] mod_sofia.c:4141 Invalid Gateway
2011-06-29 17:39:42.745992 [NOTICE] mod_sofia.c:4508 Close Channel N/A
[CS_NEW]
2011-06-29 17:39:42.745992 [NOTICE] switch_ivr_originate.c:2443 Cannot
create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
2011-06-29 17:39:42.745992 [INFO] mod_dptools.c:2680 Originate Failed.
Cause: INVALID_NUMBER_FORMAT
2011-06-29 17:39:42.745992 [NOTICE] mod_dptools.c:2799 Hangup
sofia/internal/102 at 192.168.1.76 [CS_EXECUTE] [INVALID_NUMBER_FORMAT]
2011-06-29 17:39:42.745992 [NOTICE] switch_core_session.c:1308 Session 11 (
sofia/internal/102 at 192.168.1.76) Ended
2011-06-29 17:39:42.745992 [NOTICE] switch_core_session.c:1310 Close Channel
sofia/internal/102 at 192.168.1.76 [CS_DESTROY]

the wiki doesn't make it very clear on wher to put things... :(

Thanks in advance..

Bret
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From lists at ticm.com  Wed Jun 29 17:25:04 2011
From: lists at ticm.com (Bret Watson)
Date: Wed, 29 Jun 2011 21:25:04 +0800
Subject: [Freeswitch-users] halp! newb question on setting up providers..
In-Reply-To: <BANLkTikUAUUkR0FZ=pu+9GEN9=zVhAFkaQ@mail.gmail.com>
References: <BANLkTikUAUUkR0FZ=pu+9GEN9=zVhAFkaQ@mail.gmail.com>
Message-ID: <4e0b27ca.9353e70a.2a26.1ef2@mx.google.com>

ok things are working better since I found the packt site.. 
http://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-telephony-systems/book 
much easier to understand than the wiki....

Anyway now got a strange one..

incoming calls from pennytel fail with the following..

2011-06-29 21:15:36.048108 [NOTICE] switch_channel.c:890 New Channel 
sofia/external/041xxxxxxx at 202.85.243.105 [616f79a7-763d-435b-9038-527483621a38]
2011-06-29 21:15:36.048108 [INFO] mod_dialplan_xml.c:336 Processing 
041xxxxxxx <041xxxxxxx>->618xxxxxxx in context public
2011-06-29 21:15:36.048108 [ERR] switch_core_session.c:413 Could not 
locate channel type 101 XML default
2011-06-29 21:15:36.048108 [NOTICE] switch_ivr_originate.c:2443 
Cannot create outgoing channel of type [101 XML default] cause: 
[CHAN_NOT_IMPLEMENTED]
2011-06-29 21:15:36.048108 [INFO] mod_dptools.c:2680 Originate 
Failed.  Cause: CHAN_NOT_IMPLEMENTED
2011-06-29 21:15:36.048108 [NOTICE] mod_dptools.c:2799 Hangup 
sofia/external/041xxxxxxx at 202.85.243.105 [CS_EXECUTE] [CHAN_NOT_IMPLEMENTED]
2011-06-29 21:15:36.048108 [NOTICE] switch_core_session.c:1308 
Session 1 (sofia/external/041xxxxxxx at 202.85.243.105) Ended
2011-06-29 21:15:36.048108 [NOTICE] switch_core_session.c:1310 Close 
Channel sofia/external/041xxxxxxx at 202.85.243.105 [CS_DESTROY]

The relevant dialplan  in usr/local/freeswitch/conf/dialplan/public

<include>
   <extension name="public_did">
     <condition field="destination_number" expression="^(618xxxxxxx )$">
       <action application="set" data="domain_name=$${domain}"/>
       <action application="bridge" data="101 XML default"/>
     </condition>
   </extension>
</include>

It looks pretty textbook to me, but for some reason it can't create a 
channel to that extension.. I can call that extension internally no 
problems however..

Thanks!

Bret



From wes-fs at 499x.com  Wed Jun 29 20:13:24 2011
From: wes-fs at 499x.com (Wes)
Date: Wed, 29 Jun 2011 11:13:24 -0500
Subject: [Freeswitch-users] pause and restart while recording?
In-Reply-To: <3DE0409A-DDEE-4D09-9C55-F184902EA64A@ipeva.fr>
References: <4E0A2407.3010704@499x.com>	<BANLkTi=ZDqFFmbh-xbYprMMKWQU=8HKUZg@mail.gmail.com>	<4E0B2FE0.1070801@499x.com>
	<3DE0409A-DDEE-4D09-9C55-F184902EA64A@ipeva.fr>
Message-ID: <4E0B4F24.9090404@499x.com>

if I reuse the same filename as a previous recording, it just overwrites 
it.

On 6/29/2011 10:32 AM, David Ponzone wrote:
> That's interesting, but what does FS do if you record over an existing 
> file ? Does it append or does it overwrite it ?
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service ClientIPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
> diffusion non autoris?e est interdite. Tout message ?lectronique est 
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au 
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
> n'?tes pas destinataire de ce message, merci de le d?truire 
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 29/06/2011 ? 16:00, Wes a ?crit :
>
>> I'm writing the script in LUA... not sure how to answer your question 
>> RE dialplan or ESL...
>>
>> The lua script is triggered via the socket library from a php page, 
>> and the arguments to the lua script are things like the phone number 
>> to dial, and an id number to track things back.
>>
>> The application is pretty simple, it just needs to call the desired 
>> number, play instructions via voice commands, take a recording, allow 
>> user to review the recording, rerecord it, and then finally submit 
>> it.  It's going to be for medical dictation.  It's pretty much done 
>> except for the pause feature.
>>
>> Searching around the archives for the "pause during recording" 
>> functionality, I found a few people discussing the need for pause in 
>> relation to a transcription app, but couldn't figure out what came of it.
>>
>> I'm pretty close, as I already have a loop that will stop the 
>> recording on a "#" keypress, so if it could just pause recording on a 
>> different keypress, that would be great...
>>
>> Thanks for the help!
>>
>> On 6/28/2011 6:11 PM, Michael Collins wrote:
>>> Dialplan script or ESL? A dp script is a bad idea for this kind of 
>>> thing. Anything that requires async ops like listening for digits 
>>> while doing other things is a candidate for ESL.
>>>
>>> What cool new app are you working on? :)
>>>
>>> -MC
>>>
>>> On Tue, Jun 28, 2011 at 11:57 AM, Wes <wes-fs at 499x.com 
>>> <mailto:wes-fs at 499x.com>> wrote:
>>>
>>>     I couldn't find much on this:  Is it possible in lua script to
>>>     let the
>>>     user press a key to pause recording, and press a key again to
>>>     continue
>>>     the recording, just like a traditional recorder?
>>>
>>>     Thanks!
>>>
>>>     _______________________________________________
>>>     Join us at ClueCon 2011, Aug 9-11, Chicago
>>>     http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>
>>>     FreeSWITCH-users mailing list
>>>     FreeSWITCH-users at lists.freeswitch.org
>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>>
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com  877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org 
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From msc at freeswitch.org  Wed Jun 29 20:13:30 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 29 Jun 2011 09:13:30 -0700
Subject: [Freeswitch-users] Proper prompt gain/level
In-Reply-To: <7D766BBE-F3D1-4F9C-B7B9-A44B16EB2BF6@bryansmart.com>
References: <FE081E27-8D62-4715-B7AA-64432598E013@bryansmart.com>
	<0D3AEFF37C1643B79CE3F7A7EE9EAEC3@dell9400>
	<66E81300-46E9-4C81-B2E0-123BCADDCAEB@bryansmart.com>
	<4E088A24.7010002@coppice.org>
	<3081AB39-6EF0-4C43-9AC1-7D43FCD378CD@bryansmart.com>
	<BANLkTikTK3JW=jDFNqkZqPxhic_bzL=uCQ@mail.gmail.com>
	<BANLkTimSuFpBXwANu8qc2q24z-xNYdcJnw@mail.gmail.com>
	<1BB0CF88-DA61-4AB6-9CC6-328CCB9846AD@bryansmart.com>
	<BANLkTi=9-B3R74voiHLV6RgcvvkqwS6aZQ@mail.gmail.com>
	<7D766BBE-F3D1-4F9C-B7B9-A44B16EB2BF6@bryansmart.com>
Message-ID: <BANLkTim3=9py1Pq5vVZ=hGp3j1G+ZzaSyg@mail.gmail.com>

I will give this a whirl and see what happens. I have no problems trying
that out.
-MC

On Tue, Jun 28, 2011 at 7:12 PM, Bryan Smart <bryansmart at bryansmart.com>wrote:

> Hi, Michael.
>
> Thanks for posting this.
>
> Have you considered using the higher quality sample rate conversion
> features in new versions of sox?
>
> A starting place for info is here:
> http://sox.sourceforge.net/SoX/Resampling
>
> Perhaps change:
>
> sox -v 0.2 $base_dir/$dir/$filename -r $rate -c 1
> $tmp_dir/$voice_dir/$dir/$rate/$filename
>
>
> to:
>
> sox $base_dir/$dir/$filename -c 1 $tmp_dir/$voice_dir/$dir/$rate/$filename
> rate -v -I $rate dither
>
> rate -v -I $rate: Replaces -r $rate. Uses the newer "very high quality"
> mode (instead of "high quality") with Intermediate Phase response (instead
> of linear). This setting is superior to the default, but slower to process.
>
> dither: Applies basic triangular dithering. There are other dithering
> strategies, but, based on what I've read and experienced, they aren't as
> useful when down-sampling to low rates.
>
> Might produce slightly higher quality files, the next time it is necessary
> to regenerate them.
>
> Bryan
>
> On Jun 28, 2011, at 11:18 AM, Michael Collins wrote:
>
> You can download the 48kHz files from files.freeswitch.org. The script
> itself is nothing special - all it does is cycle through the target sampling
> rates and run each file through sox. Here's a copy of the script:
>
> base_dir="48000"
> rates="48000 32000 16000 8000"
> version="1.0.16"
> voice="en-us-callie"
> voice_dir="en/us/callie"
> tar_path="../../.."
> tmp_dir="tmp"
> CWD=`pwd`
>
> for rate in $rates; do \
>   mkdir -p $tmp_dir/$voice_dir
>   for dir in `ls $base_dir`; do \
>     test -d $tmp_dir/$voice_dir/$dir/$rate || mkdir -p
> $tmp_dir/$voice_dir/$dir/$rate; \
>     for filename in `ls $base_dir/$dir`; do \
>       echo sox -v 0.2 $base_dir/$dir/$filename -r $rate -c 1
> $tmp_dir/$voice_dir/$dir/$rate/$filename; \
>       sox -v 0.2 $base_dir/$dir/$filename -r $rate -c 1
> $tmp_dir/$voice_dir/$dir/$rate/$filename; \
>     done ; \
>   done ; \
>   cd $tmp_dir
>   echo tar -cvzf
> $CWD/$tar_path/freeswitch-sounds-$voice-$rate-$version.tar.gz *; \
>   tar -cvzf $CWD/$tar_path/freeswitch-sounds-$voice-$rate-$version.tar.gz
> *; \
>   cd -
>   rm -rf $tmp_dir
> done
>
> cd $tar_path
> for rate in $rates; do \
>   openssl dgst -sha1 freeswitch-sounds-$voice-$rate-$version.tar.gz >
> freeswitch-sounds-$voice-$rate-$version.tar.gz.sha1 ; \
>   openssl dgst -md5 freeswitch-sounds-$voice-$rate-$version.tar.gz >
> freeswitch-sounds-$voice-$rate-$version.tar.gz.md5 ; \
> done
> cd $CWD
>
> Let me know if you have any suggestions.
> -MC
>
> On Mon, Jun 27, 2011 at 8:25 PM, Bryan Smart <bryansmart at bryansmart.com>wrote:
>
>> -v is a multiplier. So, -v 0.2 reduces gain to 20% of original.
>>
>> It is typical to supply professionally recorded audio with peaks boosted
>> to 0DBFS. That is probably how the prompts were originally supplied. It
>> makes since that -v 0.2 would reduce the max power to 20%, around -15 to -16
>> DBFS.
>>
>> The FS build process downloads pre-processed versions of these files. Is
>> there somewhere online that I can find the source recordings+script that
>> produces the pre-processed files that are downloaded by the build process?
>>
>> Bryan
>>
>> On Jun 27, 2011, at 7:32 PM, Michael Collins wrote:
>>
>>
>>
>> On Mon, Jun 27, 2011 at 3:07 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> Have a look at the 48khz versions of the files, those should be the
>>> originals.
>>> Someone runs a batch sox command on them to get them to the other levels.
>>>
>>
>> Just for the record, the sox command used to normalize the sounds is:
>> sox -v 0.2 $file_in -r $rate -c 1 $file_out
>>
>> The -v format is the "volume" format and if I read the sox man page
>> correctly it means a "linear amplitude adjustment". What I can't tell from
>> the man page is whether .2 means "reduce by 20%" or "reduce to 20%" or
>> something else. In any case, like Tony says, we are not picky about this as
>> long as it's not pointlessly loud.
>>
>> -MC
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Jun 29 20:14:08 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 29 Jun 2011 09:14:08 -0700
Subject: [Freeswitch-users] Proper prompt gain/level
In-Reply-To: <37721EF7-5FCD-43BF-A988-90C806E9E11F@bryansmart.com>
References: <FE081E27-8D62-4715-B7AA-64432598E013@bryansmart.com>
	<0D3AEFF37C1643B79CE3F7A7EE9EAEC3@dell9400>
	<66E81300-46E9-4C81-B2E0-123BCADDCAEB@bryansmart.com>
	<4E088A24.7010002@coppice.org>
	<3081AB39-6EF0-4C43-9AC1-7D43FCD378CD@bryansmart.com>
	<4E092ED4.60605@coppice.org>
	<2D33125A-05E0-40DE-92A2-7038940977AB@bryansmart.com>
	<4E09FA13.5030104@coppice.org>
	<88F60C8D-AFB1-46DB-9A88-4AB295A0376B@bryansmart.com>
	<BANLkTi=+DQ2h1CeS8cqkpgZ+TCnrB7=idg@mail.gmail.com>
	<37721EF7-5FCD-43BF-A988-90C806E9E11F@bryansmart.com>
Message-ID: <BANLkTima8Qa8FVzi1O0uTHYNun4ixMPKgg@mail.gmail.com>

On Tue, Jun 28, 2011 at 5:47 PM, Bryan Smart <bryansmart at bryansmart.com>wrote:

> Thanks, Anthony.
>
> I downloaded
>
>
> http://files.freeswitch.org/freeswitch-sounds-en-us-callie-48000-1.0.16.tar.gz
>
> and
>
>
> http://files.freeswitch.org/freeswitch-sounds-en-us-callie-32000-1.0.16.tar.gz
>
> I examined them with the statistical analysis tool in Sound Forge.
> Regardless of what DB scale is involved, both versions register in Sound
> Forge with identical max sample value and average RMS. This outcome doesn't
> seem possible if the 48000 set is sent through sox with "-v 0.2" to produce
> the other sets. Wouldn't the results of that processing be that the sets for
> other rates would be 20% as loud as the 48000 set?
>
> The 48000 set that I discovered must not be the correct set.
>

I'll email you offlist and see if I can't get you read-only access to our
sounds repo so you can see the raw files.
-MC
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From msc at freeswitch.org  Wed Jun 29 20:19:27 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 29 Jun 2011 09:19:27 -0700
Subject: [Freeswitch-users] FreeSWITCH Conf Call Today: Plivo,
	an OSS Twilio Alternative
Message-ID: <BANLkTims3-LbyKPuJKXAQTx_b__V3Q=EQA@mail.gmail.com>

Hello all!

Today's conference call is busy! We have Venky from the Plivo project
discussing the new software. We also have quite a few new FS items that have
been added in the last month so we need to get some documentation done. :)

Here's the agenda:

http://wiki.freeswitch.org/wiki/FS_weekly_2011_06_29

There is a link in there to a PPT for us to look at during Venky's
presentation.

Talk to you soon!
-Michael
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From infos at madovsky.org  Wed Jun 29 20:31:13 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 29 Jun 2011 12:31:13 -0400
Subject: [Freeswitch-users] change sounds file format of voicemail
Message-ID: <89F3619D8E6F437F82F3B1C00FA69725@e1705>

Where can I force voicemail to use sounds other than .wav ?

Thanks
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From steveu at coppice.org  Wed Jun 29 21:07:26 2011
From: steveu at coppice.org (Steve Underwood)
Date: Thu, 30 Jun 2011 01:07:26 +0800
Subject: [Freeswitch-users] Proper prompt gain/level
In-Reply-To: <88F60C8D-AFB1-46DB-9A88-4AB295A0376B@bryansmart.com>
References: <FE081E27-8D62-4715-B7AA-64432598E013@bryansmart.com>	<0D3AEFF37C1643B79CE3F7A7EE9EAEC3@dell9400>	<66E81300-46E9-4C81-B2E0-123BCADDCAEB@bryansmart.com>	<4E088A24.7010002@coppice.org>	<3081AB39-6EF0-4C43-9AC1-7D43FCD378CD@bryansmart.com>	<4E092ED4.60605@coppice.org>	<2D33125A-05E0-40DE-92A2-7038940977AB@bryansmart.com>	<4E09FA13.5030104@coppice.org>
	<88F60C8D-AFB1-46DB-9A88-4AB295A0376B@bryansmart.com>
Message-ID: <4E0B5BCE.7010209@coppice.org>

On 06/29/2011 03:26 AM, Bryan Smart wrote:
> I must be wrong about DBm0, then. I wasn't familiar with it, so I checked Wikipedia. Wikipedia isn't always right, of course.
>
> Wikipedia says...
> ----------
> dBm0 is an abbreviation for the power in dBm measured at a zero transmission level point.
>
> dBm0 is a concept used (amongst other areas) in audio/telephony processing since it allows a smooth integration of analog and digital chains. Notably, for
> A-law and ?-law codecs the standards define a sequence which has a 0 dBm0 output.
>
> ......
>
> Note 2: 0 dBm0 is often replaced by or used instead of digital milliwatt or zero transmission level point.
> ----------
>
> Where is my zero level transmission point? Data from a codec is not power being transmitted, only data for reproducing power at some later time. Isn't it all up to the D/A converter at the far end to determine what the power levels yielded by decoding the data will be represented relative to. If it is a hardware SIP phone, maybe the person has the volume cranked up, or turned down. Since the analog representation of the signal starts in the D/A that feeds the handset, what is considered 0? I have no idea. It sounds like this is a scale used for calibrating a D/A that feeds a pstn circuit.
Look in the G.711 spec. It defines a specific digital signal that is 
0dBm0 for both u-law and A-law.
> I admit feeling frustrated by this discussion. I've mixed music and mastered CDs for nearly 15 years, and I've always felt that 0DB in an entirely digital domain is a near universally understood concept. If I have a sample of 16-bit signed LPCM audio, then -32768 or 32767 represent the max amplitude that can be stored, and is what everyone that I know in pro audio calls 0DB when speaking strictly about a file, rather than a PA, in/out levels to an analog device like a tape machine, etc. If the gain of the encoded signal is boosted to a point where none of the samples are pushed beyond this range, then none of them clip. I thought that you'd think of DB in the same way, given we were talking about levels in files, but it feels like you're nit-picking or antagonizing me. It sounds like you're telling me that they can clip, even if the gain was never increased to a point where this range would be overflowed, but I have no idea how that is possible. It does not happen when I play such audio through the D/A on a computer's sound card, nor when I store it on a CD and play it back through a stereo.
This is highly inaccurate. Power is usually measured as an RMS value, 
and this does not relate directly to the height of the samples. It 
related to the integral of the samples over time. As I said before, the 
crest factor (basically the ratio between peak instantaneous power and 
RMS power) is about 13dB for speech. Its about the same for a voice 
singing. Its lower for most musical instruments. Nonetheless every 
musical signal has some peakiness, and 0dBOv RMS power will certainly 
clip badly. You might be refering the the measurements on PPM or other 
forms of metering which try to track the audio peaks. If those are 
working well, you can run up to 0dB on their scale before clipping. The 
RMS power will be considerably lower, though. Recording desks use peak 
power meters specifically because RMS power doesn't give you much idea 
about how close to clipping you are.
> I take your point about headroom. Still, I don't feel that the current level matches caller speech. If I felt that the prompts blended well, I wouldn't have even put myself through this thread. I've connected a mix of hardware SIP phones, desktop SIP clients, and iOS SIP clients to a conference, with no modifications to the audio level of the channel, and the level of people speaking to each other in the conference is significantly louder than any prompts that are played through it. Maybe *all* of the clients are pushing audio too strongly. I first thought it was something to do with the conference, but I soon realized that the prompts were quiet everywhere, not just when played in a conference.
Gains in the PSTN are largely controlled in a professional manner. On 
the internet its currently chaos, with many voice signal clipping 
horribly. Instead of a well balanced set of gains down the signal chain 
you get people pumping up the gain at one point, the massively 
attenuating it at another, with no regard for the distortion they are 
introducing.
> I don't expect anything to change due to my personal preference, and I realize that a background in digital audio as applies to music and live recording doesn't mean that I'm not ignorant about many things that involve digital audio as it applies to telephony. I raised the issue here to see if I might be doing something wrong. If not, I wondered if the prompt levels are set based on some sort of standard by people that are wiser regarding the details than me? It seems, though, that there really isn't a standard, and the decision is someone else's personal preference. I'd rather that personal preferences never be a default. Since there isn't an official standard/guideline, I suppose that someone has to make a decision, and that decision will be influenced by their own preferences.
Preferences shouldn't be a factor. This is supposed to be engineered.
> I have a few suggestions to help improve Michael's sox script, but that's where I'll leave this issue. It isn't worth a big argument when I can fix it myself.
>
> I apologize to the list for any smoke, hints of flames, or other frustration that might have leaked through in to my posts. The sound level thing matters to me, but it is really a small thing. I enjoy Freeswitch immensely, and really appreciate everyones' efforts in producing and updating it. I've always been fascinated with phones and any type of interactive phone app, and so Asterisk, and now Freeswitch, really spark my imagination. I'm 34, and the study of phones in my early teens was my first conceptual exposure to a large network. I phreaked a bit at the time. I was fortunate (at least in one regard) to live in the US deep south, where digital switching equipment wasn't common, and so all of the old 1970's techniques weren't unavailable. That didn't last long, but it peaked my curiosity. I used VXML for a project for an employer in 2002 or so, but didn't really feel any excitement about phones, in the way that I used to, until I ran across Asterisk. Asterisk was great for the time, but I wanted to use it more for apps than a PBX, and so was quite excited to discover the different design of Freeswitch. I rarely become excited about new environments and frameworks anymore, but Freeswitch has put me back in to a fun mood of exploration and experimentation.
There is nothing wrong with what you are doing. Its good to see someone 
care about gains.

Steve




From kris at livecall.com  Wed Jun 29 23:05:24 2011
From: kris at livecall.com (Kris)
Date: Wed, 29 Jun 2011 12:05:24 -0700
Subject: [Freeswitch-users] change sounds file format of voicemail
References: <89F3619D8E6F437F82F3B1C00FA69725@e1705>
Message-ID: <EAC9775B572D499CA7869A2FA228C517@stor1>

I am doing testing and I am about to submit changes to the GIT to allow 
8000x8 bit wav and 8000x4bit vox recording. What formats would you like to 
use?..maybe I can add them before I submit. This is my test extension and it 
works with my modifications:
Kris
<extension name="playtest">

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

<action application="answer" data=""/>

<action application="set" data="tts_engine=flite"/>

<action application="set" data="tts_voice=kal"/>

<!--

-->

<action application="set" data="record_sample_rate=8000"/>

<action application="set" data="record_format= major=WAV,subtype=PCM_U8"/>

<action application="record" data="Test1.wav 20"/>

<action application="playback" data="tone_stream://%(100,0,1004);loops=5"/>

<action application="set" data="record_format= major=WAV,subtype=PCM_16"/>

<action application="record" data="Test2.wav 20"/>

<action application="playback" data="tone_stream://%(100,0,1004);loops=5"/>

<action application="set" data="record_format= 
major=RAW,subtype=VOX_ADPCM"/>

<action application="sleep" data="200"/>

<action application="record" data="Test3.vox 20"/>

<action application="set" data="playback_delimiter=!"/>


<action application="read" data="0 1 'say:this starts at 5 seconds and plays 
for 10 at volume 
2!Test1.wav{start_ms=5000,timeout_ms=10000,speed=0,vol=2}!say:this is at 
volume -1!Test2.wav{start_ms=5000,timeout_ms=10000,speed=0,vol=-1}!say:this 
is vox!Test3.vox{start_ms=0,timeout_ms=8000,speed=2,vol=4}' digits 5000 #"/>

<action application="hangup"/>

</condition>

</extension>

----- Original Message ----- 
From: "Madovsky" <infos at madovsky.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, June 29, 2011 9:31 AM
Subject: [Freeswitch-users] change sounds file format of voicemail


Where can I force voicemail to use sounds other than .wav ?

Thanks 



From kris at livecall.com  Wed Jun 29 23:11:43 2011
From: kris at livecall.com (Kris)
Date: Wed, 29 Jun 2011 12:11:43 -0700
Subject: [Freeswitch-users] pause and restart while recording?
References: <4E0A2407.3010704@499x.com>	<BANLkTi=ZDqFFmbh-xbYprMMKWQU=8HKUZg@mail.gmail.com>	<4E0B2FE0.1070801@499x.com><3DE0409A-DDEE-4D09-9C55-F184902EA64A@ipeva.fr>
	<4E0B4F24.9090404@499x.com>
Message-ID: <1CEA045916DF425087440B38B43E905B@stor1>

i havn't tried it but it seems if you set this variable
Session.SetVariable("RECORD_APPEND", "true");

before record it should append.



Kris

----- Original Message ----- 
From: "Wes" <wes-fs at 499x.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, June 29, 2011 9:13 AM
Subject: Re: [Freeswitch-users] pause and restart while recording?


> if I reuse the same filename as a previous recording, it just overwrites
> it.
>
> On 6/29/2011 10:32 AM, David Ponzone wrote:
>> That's interesting, but what does FS do if you record over an existing
>> file ? Does it append or does it overwrite it ?
>>
>> David Ponzone Direction Technique
>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service ClientIPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com
>> <BLOCKED::http://www.ipeva-studio.com/>
>>
>> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>> ? l'intention exclusive de ses destinataires. Toute utilisation ou
>> diffusion non autoris?e est interdite. Tout message ?lectronique est
>> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au
>> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous
>> n'?tes pas destinataire de ce message, merci de le d?truire
>> imm?diatement et d'avertir l'exp?diteur./
>> /
>> /
>>
>>
>>
>> Le 29/06/2011 ? 16:00, Wes a ?crit :
>>
>>> I'm writing the script in LUA... not sure how to answer your question
>>> RE dialplan or ESL...
>>>
>>> The lua script is triggered via the socket library from a php page,
>>> and the arguments to the lua script are things like the phone number
>>> to dial, and an id number to track things back.
>>>
>>> The application is pretty simple, it just needs to call the desired
>>> number, play instructions via voice commands, take a recording, allow
>>> user to review the recording, rerecord it, and then finally submit
>>> it.  It's going to be for medical dictation.  It's pretty much done
>>> except for the pause feature.
>>>
>>> Searching around the archives for the "pause during recording"
>>> functionality, I found a few people discussing the need for pause in
>>> relation to a transcription app, but couldn't figure out what came of 
>>> it.
>>>
>>> I'm pretty close, as I already have a loop that will stop the
>>> recording on a "#" keypress, so if it could just pause recording on a
>>> different keypress, that would be great...
>>>
>>> Thanks for the help!
>>>
>>> On 6/28/2011 6:11 PM, Michael Collins wrote:
>>>> Dialplan script or ESL? A dp script is a bad idea for this kind of
>>>> thing. Anything that requires async ops like listening for digits
>>>> while doing other things is a candidate for ESL.
>>>>
>>>> What cool new app are you working on? :)
>>>>
>>>> -MC
>>>>
>>>> On Tue, Jun 28, 2011 at 11:57 AM, Wes <wes-fs at 499x.com
>>>> <mailto:wes-fs at 499x.com>> wrote:
>>>>
>>>>     I couldn't find much on this:  Is it possible in lua script to
>>>>     let the
>>>>     user press a key to pause recording, and press a key again to
>>>>     continue
>>>>     the recording, just like a traditional recorder?
>>>>
>>>>     Thanks!
>>>>
>>>>     _______________________________________________
>>>>     Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>     http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>>
>>>>     FreeSWITCH-users mailing list
>>>>     FreeSWITCH-users at lists.freeswitch.org
>>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> 
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com  877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 



From msc at freeswitch.org  Wed Jun 29 23:49:38 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 29 Jun 2011 12:49:38 -0700
Subject: [Freeswitch-users] FreeSWITCH Conf Call - Plivo Discussion -
	Recordings Up
Message-ID: <BANLkTikg4aSYo5ZH-UVq7XUBsZURt6DnWg@mail.gmail.com>

Hello all!

I knew that today's call would be popular so I did a rush job and got the
recordings up:
http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls

Just grab one of the three file types from June 29.

Enjoy!
-MC
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From jmiller at sirran.com  Wed Jun 29 23:38:04 2011
From: jmiller at sirran.com (Jim Miller)
Date: Wed, 29 Jun 2011 15:38:04 -0400
Subject: [Freeswitch-users] SRTP problem with leg B inbound to FreeSwitch
Message-ID: <4E0B7F1C.8040308@sirran.com>

I've gotten as far as I can with this problem and could use some help.

Here's the problem:
- Two soft phones can dial each other (ext 1001 and 1002) and voice 
works in both directions
- Wireshark shows Leg A in/out from phone 1 (10.1.10.5) is SRTP and Leg 
B out from freeswitch (10.1.10.2) is SRTP.  However Inbound from phone 2 
(10.1.10.4) to the switch is just RTP.

Any ideas?

I've read what I could find on the freeswitch wiki and made the obvious 
changes to the dialplan

<!-- Offer SRTP on outbound legs if we have it on inbound. -->
<action application="export" data="sip_secure_media=true"/>

and then did this to the internal.xml sofia config

<!--Uncomment to let calls hit the dialplan *before* you decide if the 
codec is ok-->
<param name="inbound-late-negotiation" value="true"/>

Using Freeswitch version 1.0.head (git-2486306 2011-06-23 11-57-38 -0500)
Blink is being used as the soft phone v0.24.1 released in May 2011. 
Linux and Mac versions.
I've also tried sflphone for a linux host with no change in the behavior.

If I bypass the media stream and send the SRTP between the soft clients 
directly all works fine.

I've attached various stripped down dialplans, sip configs, and a 
wireshark packet dump.


Thank you
-Jim
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From infos at madovsky.org  Thu Jun 30 00:15:20 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 29 Jun 2011 16:15:20 -0400
Subject: [Freeswitch-users] change sounds file format of voicemail
References: <89F3619D8E6F437F82F3B1C00FA69725@e1705>
	<EAC9775B572D499CA7869A2FA228C517@stor1>
Message-ID: <DAC98767564D4EDDA8873E1F16A5818A@e1705>

I kris,

I just converted all wav in spx, and also
G711 would be great..
do yo need I send to you some spx samples ?

Thanks
----- Original Message ----- 
From: "Kris" <kris at livecall.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, June 29, 2011 3:05 PM
Subject: Re: [Freeswitch-users] change sounds file format of voicemail


>I am doing testing and I am about to submit changes to the GIT to allow
> 8000x8 bit wav and 8000x4bit vox recording. What formats would you like to
> use?..maybe I can add them before I submit. This is my test extension and 
> it
> works with my modifications:
> Kris
> <extension name="playtest">
>
> <condition field="destination_number" expression="^playtest$">
>
> <action application="answer" data=""/>
>
> <action application="set" data="tts_engine=flite"/>
>
> <action application="set" data="tts_voice=kal"/>
>
> <!--
>
> -->
>
> <action application="set" data="record_sample_rate=8000"/>
>
> <action application="set" data="record_format= major=WAV,subtype=PCM_U8"/>
>
> <action application="record" data="Test1.wav 20"/>
>
> <action application="playback" 
> data="tone_stream://%(100,0,1004);loops=5"/>
>
> <action application="set" data="record_format= major=WAV,subtype=PCM_16"/>
>
> <action application="record" data="Test2.wav 20"/>
>
> <action application="playback" 
> data="tone_stream://%(100,0,1004);loops=5"/>
>
> <action application="set" data="record_format=
> major=RAW,subtype=VOX_ADPCM"/>
>
> <action application="sleep" data="200"/>
>
> <action application="record" data="Test3.vox 20"/>
>
> <action application="set" data="playback_delimiter=!"/>
>
>
> <action application="read" data="0 1 'say:this starts at 5 seconds and 
> plays
> for 10 at volume
> 2!Test1.wav{start_ms=5000,timeout_ms=10000,speed=0,vol=2}!say:this is at
> volume -1!Test2.wav{start_ms=5000,timeout_ms=10000,speed=0,vol=-1}!say:this
> is vox!Test3.vox{start_ms=0,timeout_ms=8000,speed=2,vol=4}' digits 5000 
> #"/>
>
> <action application="hangup"/>
>
> </condition>
>
> </extension>
>
> ----- Original Message ----- 
> From: "Madovsky" <infos at madovsky.org>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Wednesday, June 29, 2011 9:31 AM
> Subject: [Freeswitch-users] change sounds file format of voicemail
>
>
> Where can I force voicemail to use sounds other than .wav ?
>
> Thanks
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From wes-fs at 499x.com  Thu Jun 30 00:23:38 2011
From: wes-fs at 499x.com (Wes)
Date: Wed, 29 Jun 2011 15:23:38 -0500
Subject: [Freeswitch-users] pause and restart while recording?
In-Reply-To: <1CEA045916DF425087440B38B43E905B@stor1>
References: <4E0A2407.3010704@499x.com>	<BANLkTi=ZDqFFmbh-xbYprMMKWQU=8HKUZg@mail.gmail.com>	<4E0B2FE0.1070801@499x.com><3DE0409A-DDEE-4D09-9C55-F184902EA64A@ipeva.fr>	<4E0B4F24.9090404@499x.com>
	<1CEA045916DF425087440B38B43E905B@stor1>
Message-ID: <4E0B89CA.5060109@499x.com>

I tried it and got interesting results.

First, I recorded a file without this variable set.

Then, I tried recording using the same filename but with this variable 
set, and I got this error:
2011-06-29 15:07:21.284642 [ERR] mod_sndfile.c:194 Error Opening File 
[/tmp/recording.wav] [Error : Cannot open file in read/write mode due to 
string data in header.]

That made me think that when you use record_append, it will save the 
file differently than when you don't.

So I tried again, this time deleting the original wav file so it would 
create it fresh, and then, using the record_append setting, I was able 
to record my message. (speaking the number "one")

The filesize then was at:
-rw-r--r-- 1 transcribe transcribe 89052 2011-06-29 15:08 recording.wav

then, I called in again, and this time the recording should append to 
the existing recording.

I recorded my voice, and the filesize went to:
-rw-r--r-- 1 transcribe transcribe 162924 2011-06-29 15:09 recording.wav

The first message was speaking the number "one", second message was 
speaking the number "two".

Problem is, the message I heard was only the number "one"... even though 
the filesize doubled.... it did not append, and it kept the original 
message, even though it increased the filesize.

So, the record_append doesn't seem to work.  any other ideas on pausing 
and continuing recording?

On 6/29/2011 2:11 PM, Kris wrote:
> i havn't tried it but it seems if you set this variable
> Session.SetVariable("RECORD_APPEND", "true");
>
> before record it should append.
>
>
>
> Kris
>
> ----- Original Message -----
> From: "Wes"<wes-fs at 499x.com>
> To: "FreeSWITCH Users Help"<freeswitch-users at lists.freeswitch.org>
> Sent: Wednesday, June 29, 2011 9:13 AM
> Subject: Re: [Freeswitch-users] pause and restart while recording?
>
>
>> if I reuse the same filename as a previous recording, it just overwrites
>> it.
>>
>> On 6/29/2011 10:32 AM, David Ponzone wrote:
>>> That's interesting, but what does FS do if you record over an existing
>>> file ? Does it append or does it overwrite it ?
>>>
>>> David Ponzone Direction Technique
>>> email: david.ponzone at ipeva.fr<mailto:david.ponzone at ipeva.fr>
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service ClientIPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>   - www.ipeva-studio.com
>>> <BLOCKED::http://www.ipeva-studio.com/>
>>>
>>> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou
>>> diffusion non autoris?e est interdite. Tout message ?lectronique est
>>> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au
>>> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous
>>> n'?tes pas destinataire de ce message, merci de le d?truire
>>> imm?diatement et d'avertir l'exp?diteur./
>>> /
>>> /
>>>
>>>
>>>
>>> Le 29/06/2011 ? 16:00, Wes a ?crit :
>>>
>>>> I'm writing the script in LUA... not sure how to answer your question
>>>> RE dialplan or ESL...
>>>>
>>>> The lua script is triggered via the socket library from a php page,
>>>> and the arguments to the lua script are things like the phone number
>>>> to dial, and an id number to track things back.
>>>>
>>>> The application is pretty simple, it just needs to call the desired
>>>> number, play instructions via voice commands, take a recording, allow
>>>> user to review the recording, rerecord it, and then finally submit
>>>> it.  It's going to be for medical dictation.  It's pretty much done
>>>> except for the pause feature.
>>>>
>>>> Searching around the archives for the "pause during recording"
>>>> functionality, I found a few people discussing the need for pause in
>>>> relation to a transcription app, but couldn't figure out what came of
>>>> it.
>>>>
>>>> I'm pretty close, as I already have a loop that will stop the
>>>> recording on a "#" keypress, so if it could just pause recording on a
>>>> different keypress, that would be great...
>>>>
>>>> Thanks for the help!
>>>>
>>>> On 6/28/2011 6:11 PM, Michael Collins wrote:
>>>>> Dialplan script or ESL? A dp script is a bad idea for this kind of
>>>>> thing. Anything that requires async ops like listening for digits
>>>>> while doing other things is a candidate for ESL.
>>>>>
>>>>> What cool new app are you working on? :)
>>>>>
>>>>> -MC
>>>>>
>>>>> On Tue, Jun 28, 2011 at 11:57 AM, Wes<wes-fs at 499x.com
>>>>> <mailto:wes-fs at 499x.com>>  wrote:
>>>>>
>>>>>      I couldn't find much on this:  Is it possible in lua script to
>>>>>      let the
>>>>>      user press a key to pause recording, and press a key again to
>>>>>      continue
>>>>>      the recording, just like a traditional recorder?
>>>>>
>>>>>      Thanks!
>>>>>
>>>>>      _______________________________________________
>>>>>      Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>      http://www.cluecon.com<http://www.cluecon.com/>  877-7-4ACLUE
>>>>>
>>>>>      FreeSWITCH-users mailing list
>>>>>      FreeSWITCH-users at lists.freeswitch.org
>>>>>      <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>      http://www.freeswitch.org<http://www.freeswitch.org/>
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com  877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From msc at freeswitch.org  Thu Jun 30 01:12:44 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 29 Jun 2011 14:12:44 -0700
Subject: [Freeswitch-users] FreeSWITCH Cookbook - New recipes needed
Message-ID: <BANLkTikj_H49ua2KMNgAwRSReqzHXHJDsg@mail.gmail.com>

Hello all!

I've replaced the old recipe list with a new one. The link is still here on
the wiki along with other information for prospective authors:
http://wiki.freeswitch.org/wiki/Cookbook#Information_for_prospective_authors

We've made a few changes to the content. Specifically, we've removed two
chapters: "Call Center Services" and "Carrier Class Services". We do not
feel that these can properly be done in "recipe" format. So instead we've
take a few of the recipes from those chapters and put them into the "misc."
section on the FSCB recipe list. We invite your input on what to replace
them with.

What do you think? What recipes would you like to see? Send use your ideas.

Thanks,
Michael
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From gcd at i.ph  Thu Jun 30 02:38:31 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Thu, 30 Jun 2011 06:38:31 +0800
Subject: [Freeswitch-users] halp! newb question on setting up providers..
In-Reply-To: <4e0b27ca.9353e70a.2a26.1ef2@mx.google.com>
References: <BANLkTikUAUUkR0FZ=pu+9GEN9=zVhAFkaQ@mail.gmail.com>
	<4e0b27ca.9353e70a.2a26.1ef2@mx.google.com>
Message-ID: <BANLkTi=1VMnRvSLSRdQ9XR_v_7HDFu0pwQ@mail.gmail.com>

hi bret,

change:       <action application="bridge" data="101 XML default"/>
to:               <action application="transfer" data="101 XML default"/>

-nandy

On Wed, Jun 29, 2011 at 9:25 PM, Bret Watson <lists at ticm.com> wrote:

> ok things are working better since I found the packt site..
>
> http://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-telephony-systems/book
> much easier to understand than the wiki....
>
> Anyway now got a strange one..
>
> incoming calls from pennytel fail with the following..
>
> 2011-06-29 21:15:36.048108 [NOTICE] switch_channel.c:890 New Channel
> sofia/external/041xxxxxxx at 202.85.243.105[616f79a7-763d-435b-9038-527483621a38]
> 2011-06-29 21:15:36.048108 [INFO] mod_dialplan_xml.c:336 Processing
> 041xxxxxxx <041xxxxxxx>->618xxxxxxx in context public
> 2011-06-29 21:15:36.048108 [ERR] switch_core_session.c:413 Could not
> locate channel type 101 XML default
> 2011-06-29 21:15:36.048108 [NOTICE] switch_ivr_originate.c:2443
> Cannot create outgoing channel of type [101 XML default] cause:
> [CHAN_NOT_IMPLEMENTED]
> 2011-06-29 21:15:36.048108 [INFO] mod_dptools.c:2680 Originate
> Failed.  Cause: CHAN_NOT_IMPLEMENTED
> 2011-06-29 21:15:36.048108 [NOTICE] mod_dptools.c:2799 Hangup
> sofia/external/041xxxxxxx at 202.85.243.105 [CS_EXECUTE]
> [CHAN_NOT_IMPLEMENTED]
> 2011-06-29 21:15:36.048108 [NOTICE] switch_core_session.c:1308
> Session 1 (sofia/external/041xxxxxxx at 202.85.243.105) Ended
> 2011-06-29 21:15:36.048108 [NOTICE] switch_core_session.c:1310 Close
> Channel sofia/external/041xxxxxxx at 202.85.243.105 [CS_DESTROY]
>
> The relevant dialplan  in usr/local/freeswitch/conf/dialplan/public
>
> <include>
>   <extension name="public_did">
>     <condition field="destination_number" expression="^(618xxxxxxx )$">
>       <action application="set" data="domain_name=$${domain}"/>
>       <action application="bridge" data="101 XML default"/>
>     </condition>
>   </extension>
> </include>
>
> It looks pretty textbook to me, but for some reason it can't create a
> channel to that extension.. I can call that extension internally no
> problems however..
>
> Thanks!
>
> Bret
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lists at ticm.com  Thu Jun 30 05:44:39 2011
From: lists at ticm.com (Bret Watson)
Date: Thu, 30 Jun 2011 09:44:39 +0800
Subject: [Freeswitch-users] halp! newb question on setting up providers..
In-Reply-To: <BANLkTi=1VMnRvSLSRdQ9XR_v_7HDFu0pwQ@mail.gmail.com>
References: <BANLkTikUAUUkR0FZ=pu+9GEN9=zVhAFkaQ@mail.gmail.com>
	<4e0b27ca.9353e70a.2a26.1ef2@mx.google.com>
	<BANLkTi=1VMnRvSLSRdQ9XR_v_7HDFu0pwQ@mail.gmail.com>
Message-ID: <4e0bd610.8f83dc0a.3fcf.1510@mx.google.com>

works! :) thanks..

such a small tweak too..

At 06:38 AM 30/06/2011, Nandy Dagondon wrote:
>hi bret,
>
>change: ?  ?  ?  <action application="bridge" data="101 XML default"/>
>to: ?  ?  ?  ?  ?  ?  ?  <action 
>application="transfer" data="101 XML default"/>
>
>-nandy
>
>On Wed, Jun 29, 2011 at 9:25 PM, Bret Watson 
><<mailto:lists at ticm.com>lists at ticm.com> wrote:
>ok things are working better since I found the packt site..
><http://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-telephony-systems/book>http://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-telephony-systems/book
>much easier to understand than the wiki....
>
>Anyway now got a strange one..
>
>incoming calls from pennytel fail with the following..
>
>2011-06-29 21:15:36.048108 [NOTICE] switch_channel.c:890 New Channel
>sofia/external/<mailto:041xxxxxxx at 202.85.243.105>041xxxxxxx at 202.85.243.105 
>[616f79a7-763d-435b-9038-527483621a38]
>2011-06-29 21:15:36.048108 [INFO] mod_dialplan_xml.c:336 Processing
>041xxxxxxx <041xxxxxxx>->618xxxxxxx in context public
>2011-06-29 21:15:36.048108 [ERR] switch_core_session.c:413 Could not
>locate channel type 101 XML default
>2011-06-29 21:15:36.048108 [NOTICE] switch_ivr_originate.c:2443
>Cannot create outgoing channel of type [101 XML default] cause:
>[CHAN_NOT_IMPLEMENTED]
>2011-06-29 21:15:36.048108 [INFO] mod_dptools.c:2680 Originate
>Failed. ? Cause: CHAN_NOT_IMPLEMENTED
>2011-06-29 21:15:36.048108 [NOTICE] mod_dptools.c:2799 Hangup
>sofia/external/<mailto:041xxxxxxx at 202.85.243.105>041xxxxxxx at 202.85.243.105 
>[CS_EXECUTE] [CHAN_NOT_IMPLEMENTED]
>2011-06-29 21:15:36.048108 [NOTICE] switch_core_session.c:1308
>Session 1 
>(sofia/external/<mailto:041xxxxxxx at 202.85.243.105>041xxxxxxx at 202.85.243.105) 
>Ended
>2011-06-29 21:15:36.048108 [NOTICE] switch_core_session.c:1310 Close
>Channel 
>sofia/external/<mailto:041xxxxxxx at 202.85.243.105>041xxxxxxx at 202.85.243.105 
>[CS_DESTROY]
>
>The relevant dialplan ? in usr/local/freeswitch/conf/dialplan/public
>
><include>
>?  <extension name="public_did">
>?  ?  <condition field="destination_number" expression="^(618xxxxxxx )$">
>?  ?  ?  <action application="set" data="domain_name=$${domain}"/>
>?  ?  ?  <action application="bridge" data="101 XML default"/>
>?  ?  </condition>
>?  </extension>
></include>
>
>It looks pretty textbook to me, but for some reason it can't create a
>channel to that extension.. I can call that extension internally no
>problems however..
>
>Thanks!
>
>Bret
>
>
>_______________________________________________
>Join us at ClueCon 2011, Aug 9-11, Chicago
><http://www.cluecon.com>http://www.cluecon.com 877-7-4ACLUE
>
>FreeSWITCH-users mailing list
><mailto:FreeSWITCH-users at lists.freeswitch.org>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>http://lists.freeswitch.org/mailman/options/freeswitch-users
><http://www.freeswitch.org>http://www.freeswitch.org
>
>
>_______________________________________________
>Join us at ClueCon 2011, Aug 9-11, Chicago
>http://www.cluecon.com 877-7-4ACLUE
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org



From kris at livecall.com  Thu Jun 30 08:23:36 2011
From: kris at livecall.com (Kris)
Date: Wed, 29 Jun 2011 21:23:36 -0700
Subject: [Freeswitch-users] change sounds file format of voicemail
References: <89F3619D8E6F437F82F3B1C00FA69725@e1705><EAC9775B572D499CA7869A2FA228C517@stor1>
	<DAC98767564D4EDDA8873E1F16A5818A@e1705>
Message-ID: <9A9BE76A0D4E4BEBB9415FE456AE48FB@stor1>

G711 is 64kbs so my guess is to do 8000 RAW or WAV in PCM 8bits. If you send 
me the file I can try it. If you put a .wav extension on it doesn't 
Freeswitch play it?

spx may not be doable with sndfile, but I don't know it's real format..who 
knows. The supported formats are here:
http://www.mega-nerd.com/libsndfile/ and in the source sndfile.h

VOX is not listed there but it records and plays now by using RAW|VOX_APCM 
format, but that's not the only VOX format out there.

The thing is all these file extensions could have different formats so you 
have to know what you have.

Kris
email: livecallcom at hotmail.com

----- Original Message ----- 
From: "Madovsky" <infos at madovsky.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, June 29, 2011 1:15 PM
Subject: Re: [Freeswitch-users] change sounds file format of voicemail


>I kris,
>
> I just converted all wav in spx, and also
> G711 would be great..
> do yo need I send to you some spx samples ?
>
> Thanks
> ----- Original Message ----- 
> From: "Kris" <kris at livecall.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Wednesday, June 29, 2011 3:05 PM
> Subject: Re: [Freeswitch-users] change sounds file format of voicemail
>
>
>>I am doing testing and I am about to submit changes to the GIT to allow
>> 8000x8 bit wav and 8000x4bit vox recording. What formats would you like 
>> to
>> use?..maybe I can add them before I submit. This is my test extension and 
>> it
>> works with my modifications:
>> Kris
>> <extension name="playtest">
>>
>> <condition field="destination_number" expression="^playtest$">
>>
>> <action application="answer" data=""/>
>>
>> <action application="set" data="tts_engine=flite"/>
>>
>> <action application="set" data="tts_voice=kal"/>
>>
>> <!--
>>
>> -->
>>
>> <action application="set" data="record_sample_rate=8000"/>
>>
>> <action application="set" data="record_format= 
>> major=WAV,subtype=PCM_U8"/>
>>
>> <action application="record" data="Test1.wav 20"/>
>>
>> <action application="playback" 
>> data="tone_stream://%(100,0,1004);loops=5"/>
>>
>> <action application="set" data="record_format= 
>> major=WAV,subtype=PCM_16"/>
>>
>> <action application="record" data="Test2.wav 20"/>
>>
>> <action application="playback" 
>> data="tone_stream://%(100,0,1004);loops=5"/>
>>
>> <action application="set" data="record_format=
>> major=RAW,subtype=VOX_ADPCM"/>
>>
>> <action application="sleep" data="200"/>
>>
>> <action application="record" data="Test3.vox 20"/>
>>
>> <action application="set" data="playback_delimiter=!"/>
>>
>>
>> <action application="read" data="0 1 'say:this starts at 5 seconds and 
>> plays
>> for 10 at volume
>> 2!Test1.wav{start_ms=5000,timeout_ms=10000,speed=0,vol=2}!say:this is at
>> volume -1!Test2.wav{start_ms=5000,timeout_ms=10000,speed=0,vol=-1}!say:this
>> is vox!Test3.vox{start_ms=0,timeout_ms=8000,speed=2,vol=4}' digits 5000 
>> #"/>
>>
>> <action application="hangup"/>
>>
>> </condition>
>>
>> </extension>
>>
>> ----- Original Message ----- 
>> From: "Madovsky" <infos at madovsky.org>
>> To: <freeswitch-users at lists.freeswitch.org>
>> Sent: Wednesday, June 29, 2011 9:31 AM
>> Subject: [Freeswitch-users] change sounds file format of voicemail
>>
>>
>> Where can I force voicemail to use sounds other than .wav ?
>>
>> Thanks
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> 



From sharad at coraltele.com  Thu Jun 30 08:39:18 2011
From: sharad at coraltele.com (sharad)
Date: Thu, 30 Jun 2011 10:09:18 +0530
Subject: [Freeswitch-users] SIP Profile
References: <mailman.11637.1309313596.24555.freeswitch-users@lists.freeswitch.org><480E4906DC4F48E2AA9B823F5ADBCD29@sharad><549CFEF87AEDE841A38E9D15EAB4C04C59E54FA23A@cooper>
	<BANLkTimwO7jxfm1fpsNaoVaPEnohbmdYsg@mail.gmail.com>
Message-ID: <CAEAF681DDEF4DEEB8B85862819C34BD@sharad>

Thanks to all of you you provided the solution.

I think the below solution suits my requirement & it works exactly as my 
requirement is.

Thanks Marcus...

Best Regards
Sharad



----- Original Message ----- 
From: "Avi Marcus" <avi at avimarcus.net>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, June 29, 2011 6:11 PM
Subject: Re: [Freeswitch-users] SIP Profile


Yes, if you take this route, do what Peter said.. but mostly the issue is:

<X-PRE-PROCESS cmd="set" data="profile_00=192.168.4.10"/>.
<condition field="${sip_from_host}" expression="$${profile_00}">

So the match is: "it must CONTAIN the variable" - what you want is "it
must BE the variable"

To do it properly would be: <condition field="${sip_from_host}"
expression="^$${profile_00}$">

Anyway, Steven's solution is much better.
-Avi Marcus

On Wed, Jun 29, 2011 at 11:03 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
>
> You need to escape the dots, ie 192\.168\.4\.10. That should make things 
> work as expected.
>
>
>
> /Peter
>
>
>
>
>
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r sharad
> Skickat: den 29 juni 2011 09:47
> Till: freeswitch-users at lists.freeswitch.org
> ?mne: [Freeswitch-users] SIP Profile
>
>
>
> Hi,
>
> I want my FS to entertain only those calls which are coming from a 
> specific IP address .
>
> For this, I have defined a profile in vars.xml as
>
> <X-PRE-PROCESS cmd="set" data="profile_00=192.168.4.10"/>.
>
> & another line in public.xml as
>
> <extension name="transfer_to_context_01">
>
> <condition field="${sip_from_host}" expression="$${profile_00}">
>
> <action application="transfer" data="${destination_number} XML sharad"/>
>
> </condition>
>
> </extension>
>
> As per my understanding, if call is coming from 192.168.4.10, that call 
> should be processed as per the context `sharad'.....correct ?
>
> Yes, it works like this. Upto here all is ok.
>
> But if the call is coming from 192.168.100 - 192.168.4.109, these calls 
> also are processed as per context `sharad'.
>
> Means entire IP is not matched by condition field.
>
> Is there any solution for this.
>
> Requesting all of you to help me in this regard.
>
> Regards
>
> Sharad
>
> !DSPAM:4e0ada5632766217057301!
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>





From bryansmart at bryansmart.com  Thu Jun 30 08:48:32 2011
From: bryansmart at bryansmart.com (Bryan Smart)
Date: Thu, 30 Jun 2011 00:48:32 -0400
Subject: [Freeswitch-users] Public users with internal sip profiles
Message-ID: <742FA288-3479-4BAC-A6BF-277C87A39964@bryansmart.com>

I'd like both authenticated and unauthenticated users to connect to the same IP and port. I will separate them by dialplan contexts.

At the moment, only authenticated users can connect to the internal sip profile.

I have a "default" user in the directory. Its password is blank, and I've set it to use the public dialplan context. I thought that would cause unauthenticated users connecting to the internal sip profile to be authenticated as the "default" user.

Instead, the console says:

Rejected by ACL "domains". Falling back to digest auth.

This is all I'm told, even at log level 9.

I read about the acls. They seem to cause authentication based on the network address, so I thought they might be blocking connections as the "default" user. They don't seem to block connections to the external sip profile, though.

The default internal sip profile is an extremely long read. I've been through it more than a few times, but still don't understand a lot of it. I can disable authentication completely, but I need it to be optional. Are the settings that I want in the sip profile?

Thanks for any clues.

Bryan



From boris at tagnet.ru  Thu Jun 30 09:00:43 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 30 Jun 2011 11:00:43 +0600
Subject: [Freeswitch-users] SIP 100 response and media
Message-ID: <4E0C02FB.2060703@tagnet.ru>

Hello!

     I have a customers with Cisco 2821 which want to use IVR on it. He 
has configured Cisco and told me there is no voice when number is 
called. The configuration is:

Cisco 2821 <-- FreeSwitch (full proxy mode) --> PSTN

I looked at tcpdump and the call flow is:
FS (INVITE) --> Cisco 2821
Cisco 2821 (100 Trying) --> FS
Cisco 2821 (RTP) --> FS

I think the call flow is wrong in that case and SIP100 can not go 
throught Freeswitch 
(http://jdrosen.net/papers/draft-donovan-mmusic-183-00.txt) and there 
should be SIP180 to get it work. But customers said this is OK and he 
has no troubles with same configuration with other provider. What may I 
do in this sutiation? Any specific Freeswitch configuration?

P.S. FreeSWITCH Version 1.0.head (git-1c95ad9 2011-01-20 22-43-50 -0300)

-- 
Regards,
Boris




From covici at ccs.covici.com  Thu Jun 30 09:27:29 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 30 Jun 2011 01:27:29 -0400
Subject: [Freeswitch-users] Public users with internal sip profiles
In-Reply-To: <742FA288-3479-4BAC-A6BF-277C87A39964@bryansmart.com>
References: <742FA288-3479-4BAC-A6BF-277C87A39964@bryansmart.com>
Message-ID: <435.1309411649@ccs.covici.com>

Look at the public.xml and particularly the comment near the end, you
can use the same ip (5060) by default and if a user gets down to the now
commented section, and you uncomment that section, if hes authenticated
he will be transferred to the default context.

Hope that heolps.

Bryan Smart <bryansmart at bryansmart.com> wrote:

> I'd like both authenticated and unauthenticated users to connect to the same IP and port. I will separate them by dialplan contexts.
> 
> At the moment, only authenticated users can connect to the internal sip profile.
> 
> I have a "default" user in the directory. Its password is blank, and I've set it to use the public dialplan context. I thought that would cause unauthenticated users connecting to the internal sip profile to be authenticated as the "default" user.
> 
> Instead, the console says:
> 
> Rejected by ACL "domains". Falling back to digest auth.
> 
> This is all I'm told, even at log level 9.
> 
> I read about the acls. They seem to cause authentication based on the network address, so I thought they might be blocking connections as the "default" user. They don't seem to block connections to the external sip profile, though.
> 
> The default internal sip profile is an extremely long read. I've been through it more than a few times, but still don't understand a lot of it. I can disable authentication completely, but I need it to be optional. Are the settings that I want in the sip profile?
> 
> Thanks for any clues.
> 
> Bryan
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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

         John Covici
         covici at ccs.covici.com


From avi at avimarcus.net  Thu Jun 30 10:23:01 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 30 Jun 2011 09:23:01 +0300
Subject: [Freeswitch-users] NAT Traversal on SFLphone - FS not Auto Changing
	port
Message-ID: <BANLkTikOYsN8VLgHK7UTAYGTeri3nOpY_A@mail.gmail.com>

http://pastebin.freeswitch.org/16627
I've got a Linksys ATA behind NAT and a softphone behind NAT. Both seem to
register with the same UDP-NAT string and both have the same type of contact
string.. but on one FS is rewriting the RTP IP to work properly - the 1102
works, but on 1000 I don't seem to be getting any audio.

Is there some hidden parameter I can't see affecting this? Both are dialing
the same extension 9664 default MOH stuff...
This softphone actually seems to have a responsive gui in linux. If I can
get the audio to actually work, that would great!
(Oh, it has tls/srtp too, it seems)

Thanks!
<http://pastebin.freeswitch.org/16627>
-Avi Marcus
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From steveayre at gmail.com  Thu Jun 30 12:11:11 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 30 Jun 2011 09:11:11 +0100
Subject: [Freeswitch-users] SIP 100 response and media
In-Reply-To: <4E0C02FB.2060703@tagnet.ru>
References: <4E0C02FB.2060703@tagnet.ru>
Message-ID: <BANLkTikh_DOBBJQpaMLO0KBbnF-dRHEDDQ@mail.gmail.com>

Is that all the signalling?

100 Trying only acknowledges the INVITE.

That *may* be followed by 183 or 180 which'll indicate ringing. The Cisco
would only be able to send early media if that was a 183/SDP or 180/SDP.

The voice will only start when it sends 200/SDP which is when the call is
answered.

-Steve


On 30 June 2011 06:00, Boris Kovalenko <boris at tagnet.ru> wrote:

> Hello!
>
>     I have a customers with Cisco 2821 which want to use IVR on it. He
> has configured Cisco and told me there is no voice when number is
> called. The configuration is:
>
> Cisco 2821 <-- FreeSwitch (full proxy mode) --> PSTN
>
> I looked at tcpdump and the call flow is:
> FS (INVITE) --> Cisco 2821
> Cisco 2821 (100 Trying) --> FS
> Cisco 2821 (RTP) --> FS
>
> I think the call flow is wrong in that case and SIP100 can not go
> throught Freeswitch
> (http://jdrosen.net/papers/draft-donovan-mmusic-183-00.txt) and there
> should be SIP180 to get it work. But customers said this is OK and he
> has no troubles with same configuration with other provider. What may I
> do in this sutiation? Any specific Freeswitch configuration?
>
> P.S. FreeSWITCH Version 1.0.head (git-1c95ad9 2011-01-20 22-43-50 -0300)
>
> --
> Regards,
> Boris
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From boris at tagnet.ru  Thu Jun 30 12:55:29 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 30 Jun 2011 14:55:29 +0600
Subject: [Freeswitch-users] SIP 100 response and media
In-Reply-To: <BANLkTikh_DOBBJQpaMLO0KBbnF-dRHEDDQ@mail.gmail.com>
References: <4E0C02FB.2060703@tagnet.ru>
	<BANLkTikh_DOBBJQpaMLO0KBbnF-dRHEDDQ@mail.gmail.com>
Message-ID: <4E0C3A01.7040006@tagnet.ru>

Hello!

Yes, this is all. INVITE, 100 and RTP. No other SIP messages. So my 
knowledge of SIP if right. And customer lies to me or forgot to 
configure something :)

> Is that all the signalling?
>
> 100 Trying only acknowledges the INVITE.
>
> That *may* be followed by 183 or 180 which'll indicate ringing. The 
> Cisco would only be able to send early media if that was a 183/SDP or 
> 180/SDP.
>
> The voice will only start when it sends 200/SDP which is when the call 
> is answered.
>
> -Steve
>
>
> On 30 June 2011 06:00, Boris Kovalenko <boris at tagnet.ru 
> <mailto:boris at tagnet.ru>> wrote:
>
>     Hello!
>
>         I have a customers with Cisco 2821 which want to use IVR on it. He
>     has configured Cisco and told me there is no voice when number is
>     called. The configuration is:
>
>     Cisco 2821 <-- FreeSwitch (full proxy mode) --> PSTN
>
>     I looked at tcpdump and the call flow is:
>     FS (INVITE) --> Cisco 2821
>     Cisco 2821 (100 Trying) --> FS
>     Cisco 2821 (RTP) --> FS
>
>     I think the call flow is wrong in that case and SIP100 can not go
>     throught Freeswitch
>     (http://jdrosen.net/papers/draft-donovan-mmusic-183-00.txt) and there
>     should be SIP180 to get it work. But customers said this is OK and he
>     has no troubles with same configuration with other provider. What
>     may I
>     do in this sutiation? Any specific Freeswitch configuration?
>
>     P.S. FreeSWITCH Version 1.0.head (git-1c95ad9 2011-01-20 22-43-50
>     -0300)
>
>     --
>     Regards,
>     Boris
>
>
>
>     _______________________________________________
>     Join us at ClueCon 2011, Aug 9-11, Chicago
>     http://www.cluecon.com 877-7-4ACLUE
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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

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From steveayre at gmail.com  Thu Jun 30 15:32:14 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 30 Jun 2011 12:32:14 +0100
Subject: [Freeswitch-users] SIP 100 response and media
In-Reply-To: <4E0C3A01.7040006@tagnet.ru>
References: <4E0C02FB.2060703@tagnet.ru>
	<BANLkTikh_DOBBJQpaMLO0KBbnF-dRHEDDQ@mail.gmail.com>
	<4E0C3A01.7040006@tagnet.ru>
Message-ID: <BANLkTinNi0=5G00vmRaPZmj-WnVDOAxKYc+6CwsuZgYF_cdigQ@mail.gmail.com>

Btw, this is the latest SIP spec, better than the draft spec for 183 you
posted before.

http://www.ietf.org/rfc/rfc3261.txt

-Steve



On 30 June 2011 09:55, Boris Kovalenko <boris at tagnet.ru> wrote:

> **
> Hello!
>
> Yes, this is all. INVITE, 100 and RTP. No other SIP messages. So my
> knowledge of SIP if right. And customer lies to me or forgot to configure
> something :)
>
>
> Is that all the signalling?
>
> 100 Trying only acknowledges the INVITE.
>
> That *may* be followed by 183 or 180 which'll indicate ringing. The Cisco
> would only be able to send early media if that was a 183/SDP or 180/SDP.
>
> The voice will only start when it sends 200/SDP which is when the call is
> answered.
>
> -Steve
>
>
> On 30 June 2011 06:00, Boris Kovalenko <boris at tagnet.ru> wrote:
>
>> Hello!
>>
>>     I have a customers with Cisco 2821 which want to use IVR on it. He
>> has configured Cisco and told me there is no voice when number is
>> called. The configuration is:
>>
>> Cisco 2821 <-- FreeSwitch (full proxy mode) --> PSTN
>>
>> I looked at tcpdump and the call flow is:
>> FS (INVITE) --> Cisco 2821
>> Cisco 2821 (100 Trying) --> FS
>> Cisco 2821 (RTP) --> FS
>>
>> I think the call flow is wrong in that case and SIP100 can not go
>> throught Freeswitch
>> (http://jdrosen.net/papers/draft-donovan-mmusic-183-00.txt) and there
>> should be SIP180 to get it work. But customers said this is OK and he
>> has no troubles with same configuration with other provider. What may I
>> do in this sutiation? Any specific Freeswitch configuration?
>>
>> P.S. FreeSWITCH Version 1.0.head (git-1c95ad9 2011-01-20 22-43-50 -0300)
>>
>> --
>> Regards,
>> Boris
>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> ? ?????????,
>   ????? ?????????
>   ??? "??????"
>   ???. +7 (3435) 230001
>   ???? +7 (3435) 230005
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From boris at tagnet.ru  Thu Jun 30 16:00:02 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 30 Jun 2011 18:00:02 +0600
Subject: [Freeswitch-users] SIP 100 response and media
In-Reply-To: <BANLkTinNi0=5G00vmRaPZmj-WnVDOAxKYc+6CwsuZgYF_cdigQ@mail.gmail.com>
References: <4E0C02FB.2060703@tagnet.ru>	<BANLkTikh_DOBBJQpaMLO0KBbnF-dRHEDDQ@mail.gmail.com>	<4E0C3A01.7040006@tagnet.ru>
	<BANLkTinNi0=5G00vmRaPZmj-WnVDOAxKYc+6CwsuZgYF_cdigQ@mail.gmail.com>
Message-ID: <4E0C6542.1040206@tagnet.ru>

Hello!

     But even reading this RFC, I found that 100 Trying should be 
followed by one of 18x response to get media channels connected, isn't?

> Btw, this is the latest SIP spec, better than the draft spec for 183 
> you posted before.
>
> http://www.ietf.org/rfc/rfc3261.txt
>
> -Steve
>
>
>
> On 30 June 2011 09:55, Boris Kovalenko <boris at tagnet.ru 
> <mailto:boris at tagnet.ru>> wrote:
>
>     Hello!
>
>     Yes, this is all. INVITE, 100 and RTP. No other SIP messages. So
>     my knowledge of SIP if right. And customer lies to me or forgot to
>     configure something :)
>
>
>>     Is that all the signalling?
>>
>>     100 Trying only acknowledges the INVITE.
>>
>>     That *may* be followed by 183 or 180 which'll indicate ringing.
>>     The Cisco would only be able to send early media if that was a
>>     183/SDP or 180/SDP.
>>
>>     The voice will only start when it sends 200/SDP which is when the
>>     call is answered.
>>
>>     -Steve
>>
>>
>>     On 30 June 2011 06:00, Boris Kovalenko <boris at tagnet.ru
>>     <mailto:boris at tagnet.ru>> wrote:
>>
>>         Hello!
>>
>>             I have a customers with Cisco 2821 which want to use IVR
>>         on it. He
>>         has configured Cisco and told me there is no voice when number is
>>         called. The configuration is:
>>
>>         Cisco 2821 <-- FreeSwitch (full proxy mode) --> PSTN
>>
>>         I looked at tcpdump and the call flow is:
>>         FS (INVITE) --> Cisco 2821
>>         Cisco 2821 (100 Trying) --> FS
>>         Cisco 2821 (RTP) --> FS
>>
>>         I think the call flow is wrong in that case and SIP100 can not go
>>         throught Freeswitch
>>         (http://jdrosen.net/papers/draft-donovan-mmusic-183-00.txt)
>>         and there
>>         should be SIP180 to get it work. But customers said this is
>>         OK and he
>>         has no troubles with same configuration with other provider.
>>         What may I
>>         do in this sutiation? Any specific Freeswitch configuration?
>>
>>         P.S. FreeSWITCH Version 1.0.head (git-1c95ad9 2011-01-20
>>         22-43-50 -0300)
>>
>>         --
>>         Regards,
>>         Boris
>>
>>
>>
>>         _______________________________________________
>>         Join us at ClueCon 2011, Aug 9-11, Chicago
>>         http://www.cluecon.com 877-7-4ACLUE
>>
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>     _______________________________________________
>>     Join us at ClueCon 2011, Aug 9-11, Chicago
>>     http://www.cluecon.com  877-7-4ACLUE
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>     -- 
>     ? ?????????,
>        ????? ?????????
>        ??? "??????"
>        ???. +7 (3435) 230001
>        ???? +7 (3435) 230005
>
>
>     _______________________________________________
>     Join us at ClueCon 2011, Aug 9-11, Chicago
>     http://www.cluecon.com 877-7-4ACLUE
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Regards,
Boris


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From covici at ccs.covici.com  Thu Jun 30 16:27:07 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 30 Jun 2011 08:27:07 -0400
Subject: [Freeswitch-users] segfault after 407 response
Message-ID: <23273.1309436827@ccs.covici.com>

Hi.  I have an fs install from about a week ago, and I am using an
extension on the external sip port of 5060.  Its default context is
public and I have uncommented the last section where it asks for a sip
authorization.  When it receives the 407 challenge, I am getting a seg
fault.  I have traces, but they are with the default configs, so a lot
of things are optimized out.  Should I do them again, file a bug, or is
this something which is known?

Thanks in advance for any suggestions.

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

         John Covici
         covici at ccs.covici.com


From boris at tagnet.ru  Thu Jun 30 16:27:07 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 30 Jun 2011 18:27:07 +0600
Subject: [Freeswitch-users] SIP 100 response and media
In-Reply-To: <BANLkTinNi0=5G00vmRaPZmj-WnVDOAxKYc+6CwsuZgYF_cdigQ@mail.gmail.com>
References: <4E0C02FB.2060703@tagnet.ru>	<BANLkTikh_DOBBJQpaMLO0KBbnF-dRHEDDQ@mail.gmail.com>	<4E0C3A01.7040006@tagnet.ru>
	<BANLkTinNi0=5G00vmRaPZmj-WnVDOAxKYc+6CwsuZgYF_cdigQ@mail.gmail.com>
Message-ID: <4E0C6B9B.8080604@tagnet.ru>

Hello!

     Have reread your post. So 100 Trying may be (or may be not) 
followed by 180/183. And in case of 180/183 early media will be 
processed (one way only?). And full media will be only after the 200 OK 
from Cisco side. It this right? In other words - Cisco *MUST* respond 
with 100 Trying and 200 OK at least?


> Btw, this is the latest SIP spec, better than the draft spec for 183 
> you posted before.
>
> http://www.ietf.org/rfc/rfc3261.txt
>
> -Steve
>
>
>
> On 30 June 2011 09:55, Boris Kovalenko <boris at tagnet.ru 
> <mailto:boris at tagnet.ru>> wrote:
>
>     Hello!
>
>     Yes, this is all. INVITE, 100 and RTP. No other SIP messages. So
>     my knowledge of SIP if right. And customer lies to me or forgot to
>     configure something :)
>
>
>>     Is that all the signalling?
>>
>>     100 Trying only acknowledges the INVITE.
>>
>>     That *may* be followed by 183 or 180 which'll indicate ringing.
>>     The Cisco would only be able to send early media if that was a
>>     183/SDP or 180/SDP.
>>
>>     The voice will only start when it sends 200/SDP which is when the
>>     call is answered.
>>
>>     -Steve
>>
>>
>>     On 30 June 2011 06:00, Boris Kovalenko <boris at tagnet.ru
>>     <mailto:boris at tagnet.ru>> wrote:
>>
>>         Hello!
>>
>>             I have a customers with Cisco 2821 which want to use IVR
>>         on it. He
>>         has configured Cisco and told me there is no voice when number is
>>         called. The configuration is:
>>
>>         Cisco 2821 <-- FreeSwitch (full proxy mode) --> PSTN
>>
>>         I looked at tcpdump and the call flow is:
>>         FS (INVITE) --> Cisco 2821
>>         Cisco 2821 (100 Trying) --> FS
>>         Cisco 2821 (RTP) --> FS
>>
>>         I think the call flow is wrong in that case and SIP100 can not go
>>         throught Freeswitch
>>         (http://jdrosen.net/papers/draft-donovan-mmusic-183-00.txt)
>>         and there
>>         should be SIP180 to get it work. But customers said this is
>>         OK and he
>>         has no troubles with same configuration with other provider.
>>         What may I
>>         do in this sutiation? Any specific Freeswitch configuration?
>>
>>         P.S. FreeSWITCH Version 1.0.head (git-1c95ad9 2011-01-20
>>         22-43-50 -0300)
>>
>>         --
>>         Regards,
>>         Boris
>>
>>
>>
>>         _______________________________________________
>>         Join us at ClueCon 2011, Aug 9-11, Chicago
>>         http://www.cluecon.com 877-7-4ACLUE
>>
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>     _______________________________________________
>>     Join us at ClueCon 2011, Aug 9-11, Chicago
>>     http://www.cluecon.com  877-7-4ACLUE
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>     -- 
>     ? ?????????,
>        ????? ?????????
>        ??? "??????"
>        ???. +7 (3435) 230001
>        ???? +7 (3435) 230005
>
>
>     _______________________________________________
>     Join us at ClueCon 2011, Aug 9-11, Chicago
>     http://www.cluecon.com 877-7-4ACLUE
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Regards,
Boris


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From kerem.erciyes at gmail.com  Thu Jun 30 16:30:11 2011
From: kerem.erciyes at gmail.com (Kerem Erciyes)
Date: Thu, 30 Jun 2011 15:30:11 +0300
Subject: [Freeswitch-users] SIP 100 response and media
In-Reply-To: <4E0C6B9B.8080604@tagnet.ru>
References: <4E0C02FB.2060703@tagnet.ru>
	<BANLkTikh_DOBBJQpaMLO0KBbnF-dRHEDDQ@mail.gmail.com>
	<4E0C3A01.7040006@tagnet.ru>
	<BANLkTinNi0=5G00vmRaPZmj-WnVDOAxKYc+6CwsuZgYF_cdigQ@mail.gmail.com>
	<4E0C6B9B.8080604@tagnet.ru>
Message-ID: <BANLkTimMMRH9d2krF+TyAPy=9TkMRT1uNA@mail.gmail.com>

Hi Boris,

I had some trouble with 28xx series before, but I was trying to use it
as a PSTN-2-SIP gateway. Try to see if the firmware is up-to-date, as
there were some SIP protocol related glithces in the code fixed in
newer firmware.

Kerem


On Thu, Jun 30, 2011 at 3:27 PM, Boris Kovalenko <boris at tagnet.ru> wrote:
> Hello!
>
> ??? Have reread your post. So 100 Trying may be (or may be not) followed by
> 180/183. And in case of 180/183 early media will be processed (one way
> only?). And full media will be only after the 200 OK from Cisco side. It
> this right? In other words - Cisco *MUST* respond with 100 Trying and 200 OK
> at least?
>
>
> Btw, this is the latest SIP spec, better than the draft spec for 183 you
> posted before.
>
> http://www.ietf.org/rfc/rfc3261.txt
>
> -Steve
>
>
>
> On 30 June 2011 09:55, Boris Kovalenko <boris at tagnet.ru> wrote:
>>
>> Hello!
>>
>> Yes, this is all. INVITE, 100 and RTP. No other SIP messages. So my
>> knowledge of SIP if right. And customer lies to me or forgot to configure
>> something :)
>>
>> Is that all the signalling?
>>
>> 100 Trying only acknowledges the INVITE.
>>
>> That *may* be followed by 183 or 180 which'll indicate ringing. The Cisco
>> would only be able to send early media if that was a 183/SDP or 180/SDP.
>>
>> The voice will only start when it sends 200/SDP which is when the call is
>> answered.
>>
>> -Steve
>>
>>
>> On 30 June 2011 06:00, Boris Kovalenko <boris at tagnet.ru> wrote:
>>>
>>> Hello!
>>>
>>> ? ? I have a customers with Cisco 2821 which want to use IVR on it. He
>>> has configured Cisco and told me there is no voice when number is
>>> called. The configuration is:
>>>
>>> Cisco 2821 <-- FreeSwitch (full proxy mode) --> PSTN
>>>
>>> I looked at tcpdump and the call flow is:
>>> FS (INVITE) --> Cisco 2821
>>> Cisco 2821 (100 Trying) --> FS
>>> Cisco 2821 (RTP) --> FS
>>>
>>> I think the call flow is wrong in that case and SIP100 can not go
>>> throught Freeswitch
>>> (http://jdrosen.net/papers/draft-donovan-mmusic-183-00.txt) and there
>>> should be SIP180 to get it work. But customers said this is OK and he
>>> has no troubles with same configuration with other provider. What may I
>>> do in this sutiation? Any specific Freeswitch configuration?
>>>
>>> P.S. FreeSWITCH Version 1.0.head (git-1c95ad9 2011-01-20 22-43-50 -0300)
>>>
>>> --
>>> Regards,
>>> Boris
>>>
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> --
>> ? ?????????,
>>   ????? ?????????
>>   ??? "??????"
>>   ???. +7 (3435) 230001
>>   ???? +7 (3435) 230005
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> --
> Regards,
> Boris
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kerem Erciyes - Sistem Danismani
http://keremerciyes.com


From steveayre at gmail.com  Thu Jun 30 16:51:18 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 30 Jun 2011 13:51:18 +0100
Subject: [Freeswitch-users] SIP 100 response and media
In-Reply-To: <4E0C6542.1040206@tagnet.ru>
References: <4E0C02FB.2060703@tagnet.ru>
	<BANLkTikh_DOBBJQpaMLO0KBbnF-dRHEDDQ@mail.gmail.com>
	<4E0C3A01.7040006@tagnet.ru>
	<BANLkTinNi0=5G00vmRaPZmj-WnVDOAxKYc+6CwsuZgYF_cdigQ@mail.gmail.com>
	<4E0C6542.1040206@tagnet.ru>
Message-ID: <BANLkTim4hoBnmSXjRDyqqU4z7YzQ7s1CnQ@mail.gmail.com>

Yes, either that or 200 (you can skip ringing to go straight to being
answered).

Generally with 183/180 you'll get ringback but not hear voice in early
media, you wouldn't normally hear voice until the 200 OK answers the call.

Media can't start up without SDP in the body of the 180/183/200. SDP
contains the IP/port used for media and the selected codec. Media can't work
without those details. Sometimes you'll see 180/183 without SDP there'll
still be no early media containing ringback, in that case the client'll
usually generate its own ringback but you'll always see SDP in the 200 OK.

-Steve




On 30 June 2011 13:00, Boris Kovalenko <boris at tagnet.ru> wrote:

> **
> Hello!
>
>     But even reading this RFC, I found that 100 Trying should be followed
> by one of 18x response to get media channels connected, isn't?
>
> Btw, this is the latest SIP spec, better than the draft spec for 183 you
> posted before.
>
> http://www.ietf.org/rfc/rfc3261.txt
>
> -Steve
>
>
>
>  On 30 June 2011 09:55, Boris Kovalenko <boris at tagnet.ru> wrote:
>
>>  Hello!
>>
>> Yes, this is all. INVITE, 100 and RTP. No other SIP messages. So my
>> knowledge of SIP if right. And customer lies to me or forgot to configure
>> something :)
>>
>>
>> Is that all the signalling?
>>
>> 100 Trying only acknowledges the INVITE.
>>
>> That *may* be followed by 183 or 180 which'll indicate ringing. The Cisco
>> would only be able to send early media if that was a 183/SDP or 180/SDP.
>>
>> The voice will only start when it sends 200/SDP which is when the call is
>> answered.
>>
>> -Steve
>>
>>
>> On 30 June 2011 06:00, Boris Kovalenko <boris at tagnet.ru> wrote:
>>
>>> Hello!
>>>
>>>     I have a customers with Cisco 2821 which want to use IVR on it. He
>>> has configured Cisco and told me there is no voice when number is
>>> called. The configuration is:
>>>
>>> Cisco 2821 <-- FreeSwitch (full proxy mode) --> PSTN
>>>
>>> I looked at tcpdump and the call flow is:
>>> FS (INVITE) --> Cisco 2821
>>> Cisco 2821 (100 Trying) --> FS
>>> Cisco 2821 (RTP) --> FS
>>>
>>> I think the call flow is wrong in that case and SIP100 can not go
>>> throught Freeswitch
>>> (http://jdrosen.net/papers/draft-donovan-mmusic-183-00.txt) and there
>>> should be SIP180 to get it work. But customers said this is OK and he
>>> has no troubles with same configuration with other provider. What may I
>>> do in this sutiation? Any specific Freeswitch configuration?
>>>
>>> P.S. FreeSWITCH Version 1.0.head (git-1c95ad9 2011-01-20 22-43-50 -0300)
>>>
>>> --
>>> Regards,
>>> Boris
>>>
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>>  --
>> ? ?????????,
>>   ????? ?????????
>>   ??? "??????"
>>   ???. +7 (3435) 230001
>>   ???? +7 (3435) 230005
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> Regards,
> Boris
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Thu Jun 30 16:53:01 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 30 Jun 2011 13:53:01 +0100
Subject: [Freeswitch-users] segfault after 407 response
In-Reply-To: <23273.1309436827@ccs.covici.com>
References: <23273.1309436827@ccs.covici.com>
Message-ID: <BANLkTi=gMMMaLico_5DdCYzEO875+cAH2g@mail.gmail.com>

Can you reproduce it on the latest Git? If so, get a coredump and file a
Jira.

-Steve



On 30 June 2011 13:27, <covici at ccs.covici.com> wrote:

> Hi.  I have an fs install from about a week ago, and I am using an
> extension on the external sip port of 5060.  Its default context is r
> public and I have uncommented the last section where it asks for a sip
> authorization.  When it receives the 407 challenge, I am getting a seg
> fault.  I have traces, but they are with the default configs, so a lot
> of things are optimized out.  Should I do them again, file a bug, or is
> this something which is known?
>
> Thanks in advance for any suggestions.
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.com
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ankitwalia4u at gmail.com  Thu Jun 30 17:37:56 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Thu, 30 Jun 2011 19:07:56 +0530
Subject: [Freeswitch-users] Could not connect thru GSMOpen
Message-ID: <BANLkTimk9mc9M66kE0_h5bF5bKf2S0NReg@mail.gmail.com>

Hi all,

I have properly compiled the gsmopen module.

Added the Dialplan

    <extension name="gsmopen">
      <condition field="destination_number" expression="^1002$">
        <action application="bridge" data="gsmopen/interface1/xxxxxxxxx"/>
      </condition>
    </extension>
----------------------------------------------------
And the gsm conf file
<configuration name="gsmopen.conf" description="GSMopen Configuration">
-
<global_settings>
<param name="debug" value="8"/>
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
<param name="hold-music" value="$${moh_uri}"/>
<param name="destination" value="9999"/>
</global_settings>
<!-- one entry here per gsmopen interface -->
-
<per_interface_settings>
-
<interface id="1" name="interface1">
<param name="hold-music" value="$${moh_uri}"/>
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
<param name="destination" value="5000"/>
<param name="alsacname" value="plughw:1"/>
<param name="alsapname" value="plughw:1"/>
<param name="controldevice_name" value="/dev/ttyACM1"/>
</interface>
<!-- following interfaces are commented out!
    -->
</per_interface_settings>
</configuration>
------------------------------------------------------------------
I am using a Nokia handset in India
---------------------------------------------------------------------
I am getting the below error on gsm_load.

gsm reload
2011-06-30 19:03:52.716491 [WARNING] mod_gsmopen.cpp:1860 rev
exported[(nil)|37     ][WARNINGA  1860 ][interface1][-1, 0, 0]
STARTING interface_id=1
2011-06-30 19:03:56.616460 [ERR] gsmopen_protocol.cpp:2999 rev
exported[(nil)|37     ][ERRORA  2999 ][interface1][-1, 0, 0]
snd_pcm_open failed with error 'No such file or directory' on device
'plughw:1', if you are using a plughw:n device please change it to be
a default:n device (so to allow it to be shared with other concurrent
programs), or maybe you are using an ALSA voicemodem and slmodemd is
running?
2011-06-30 19:03:56.616460 [ERR] gsmopen_protocol.cpp:2891 rev
exported[(nil)|37     ][ERRORA  2891 ][interface1][-1, 0, 0] Failed
opening ALSA capture device: plughw:1
2011-06-30 19:03:56.616460 [ERR] mod_gsmopen.cpp:1931 rev
exported[(nil)|37     ][ERRORA  1931 ][interface1][-1, 0, 0] alsa_init
failed
2011-06-30 19:03:56.616460 [ERR] mod_gsmopen.cpp:1932 rev
exported[(nil)|37     ][ERRORA  1932 ][interface1][-1, 0, 0] STARTING
interface_id=1 FAILED
2011-06-30 19:03:56.616460 [ERR] mod_gsmopen.cpp:3146 rev
exported[(nil)|37     ][ERRORA  3146 ][interface1][-1, 0, 0] ALARM on
interface interface1:

-----------------------------------------------------------------------------------------------------------------
I could not understand what does the error message trying to say? I am
connected to friend's mobile with datacable on com1 port. I jsut wanted to
check SMS. Please help.

Thanks
Ankit

-- 
Life is like a rose its upto u feel it as its fragrance or thorns
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From covici at ccs.covici.com  Thu Jun 30 18:10:42 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 30 Jun 2011 10:10:42 -0400
Subject: [Freeswitch-users] segfault after 407 response
In-Reply-To: <BANLkTi=gMMMaLico_5DdCYzEO875+cAH2g@mail.gmail.com>
References: <23273.1309436827@ccs.covici.com>
	<BANLkTi=gMMMaLico_5DdCYzEO875+cAH2g@mail.gmail.com>
Message-ID: <4524.1309443042@ccs.covici.com>

I could not get a core dump, but I can run under gdb and get traces, but
I have to compile without optimization and with -g gdb, so are the
traces of any use without that?

Steven Ayre <steveayre at gmail.com> wrote:

> Can you reproduce it on the latest Git? If so, get a coredump and file a
> Jira.
> 
> -Steve
> 
> 
> 
> On 30 June 2011 13:27, <covici at ccs.covici.com> wrote:
> 
> > Hi.  I have an fs install from about a week ago, and I am using an
> > extension on the external sip port of 5060.  Its default context is r
> > public and I have uncommented the last section where it asks for a sip
> > authorization.  When it receives the 407 challenge, I am getting a seg
> > fault.  I have traces, but they are with the default configs, so a lot
> > of things are optimized out.  Should I do them again, file a bug, or is
> > this something which is known?
> >
> > Thanks in advance for any suggestions.
> >
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> >
> >         John Covici
> >         covici at ccs.covici.com
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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

         John Covici
         covici at ccs.covici.com


From gmaruzz at gmail.com  Thu Jun 30 18:04:54 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 30 Jun 2011 16:04:54 +0200
Subject: [Freeswitch-users] Could not connect thru GSMOpen
In-Reply-To: <BANLkTimk9mc9M66kE0_h5bF5bKf2S0NReg@mail.gmail.com>
References: <BANLkTimk9mc9M66kE0_h5bF5bKf2S0NReg@mail.gmail.com>
Message-ID: <BANLkTimkhpspbUu+JiqNTrWtSZaRvRdWJg@mail.gmail.com>

the "plughw:1" alsa device does not exists (or is not read-write to
the FS user).

Are you using a soundcard connected to the cellphone? To be audio
enabled, gsmopen need a soundcard connected to the cellphone (as
explained in the wiki page).

If you're not interested in audio, but only in SMS, then compile
gsmopen without audio, so it will not check for the soundcard.

Also, I don't know if gsmopen will work with the kind of cellphone
you're using. For sure it works with mobigater embedded devices, with
motorolas, with ericssons.

-giovanni
-giovanni

On Thu, Jun 30, 2011 at 3:37 PM, ankIT WALiA <ankitwalia4u at gmail.com> wrote:
> Hi all,
>
> I have properly compiled the gsmopen module.
>
> Added the Dialplan
>
>     <extension name="gsmopen">
>       <condition field="destination_number" expression="^1002$">
>         <action application="bridge" data="gsmopen/interface1/xxxxxxxxx"/>
>       </condition>
>     </extension>
> ----------------------------------------------------
> And the gsm conf file
> <configuration name="gsmopen.conf" description="GSMopen Configuration">
> -
> <global_settings>
>
> <param name="debug" value="8"/>
> <param name="dialplan" value="XML"/>
> <param name="context" value="default"/>
> <param name="hold-music" value="$${moh_uri}"/>
>
> <param name="destination" value="9999"/>
> </global_settings>
> <!-- one entry here per gsmopen interface -->
> -
> <per_interface_settings>
> -
> <interface id="1" name="interface1">
>
> <param name="hold-music" value="$${moh_uri}"/>
> <param name="dialplan" value="XML"/>
> <param name="context" value="default"/>
> <param name="destination" value="5000"/>
>
> <param name="alsacname" value="plughw:1"/>
> <param name="alsapname" value="plughw:1"/>
> <param name="controldevice_name" value="/dev/ttyACM1"/>
>
> </interface>
> <!-- following interfaces are commented out!
>     -->
> </per_interface_settings>
> </configuration>
> ------------------------------------------------------------------
> I am using a Nokia handset in India
>
> ---------------------------------------------------------------------
> I am getting the below error on gsm_load.
>
> gsm reload
> 2011-06-30 19:03:52.716491 [WARNING] mod_gsmopen.cpp:1860 rev
> exported[(nil)|37     ][WARNINGA  1860 ][interface1][-1, 0, 0] STARTING
> interface_id=1
>
> 2011-06-30 19:03:56.616460 [ERR] gsmopen_protocol.cpp:2999 rev
> exported[(nil)|37     ][ERRORA  2999 ][interface1][-1, 0, 0] snd_pcm_open
> failed with error 'No such file or directory' on device 'plughw:1', if you
> are using a plughw:n device please change it to be a default:n device (so to
> allow it to be shared with other concurrent programs), or maybe you are
> using an ALSA voicemodem and slmodemd is running?
>
> 2011-06-30 19:03:56.616460 [ERR] gsmopen_protocol.cpp:2891 rev
> exported[(nil)|37     ][ERRORA  2891 ][interface1][-1, 0, 0] Failed opening
> ALSA capture device: plughw:1
> 2011-06-30 19:03:56.616460 [ERR] mod_gsmopen.cpp:1931 rev exported[(nil)|37
> ][ERRORA  1931 ][interface1][-1, 0, 0] alsa_init failed
>
> 2011-06-30 19:03:56.616460 [ERR] mod_gsmopen.cpp:1932 rev exported[(nil)|37
> ][ERRORA  1932 ][interface1][-1, 0, 0] STARTING interface_id=1 FAILED
> 2011-06-30 19:03:56.616460 [ERR] mod_gsmopen.cpp:3146 rev exported[(nil)|37
> ][ERRORA  3146 ][interface1][-1, 0, 0] ALARM on interface interface1:
>
>
> -----------------------------------------------------------------------------------------------------------------
> I could not understand what does the error message trying to say? I am
> connected to friend's mobile with datacable on com1 port. I jsut wanted to
> check SMS. Please help.
>
> Thanks
> Ankit
>
> --
> Life is like a rose its upto u feel it as its fragrance or thorns
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From brian at freeswitch.org  Thu Jun 30 18:14:36 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 30 Jun 2011 10:14:36 -0400
Subject: [Freeswitch-users] segfault after 407 response
In-Reply-To: <4524.1309443042@ccs.covici.com>
References: <23273.1309436827@ccs.covici.com>
	<BANLkTi=gMMMaLico_5DdCYzEO875+cAH2g@mail.gmail.com>
	<4524.1309443042@ccs.covici.com>
Message-ID: <4832486C-1E5B-47F0-90BB-0C86CF41C2F3@freeswitch.org>

Unless you did something you can dump core with our default with 'ulimit -c unlimited' because we do NOT compile optimized by default.

/b

On Jun 30, 2011, at 10:10 AM, covici at ccs.covici.com wrote:

> I could not get a core dump, but I can run under gdb and get traces, but
> I have to compile without optimization and with -g gdb, so are the
> traces of any use without that?



From boris at tagnet.ru  Thu Jun 30 18:44:35 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 30 Jun 2011 20:44:35 +0600
Subject: [Freeswitch-users] SIP 100 response and media
In-Reply-To: <BANLkTimMMRH9d2krF+TyAPy=9TkMRT1uNA@mail.gmail.com>
References: <4E0C02FB.2060703@tagnet.ru>	<BANLkTikh_DOBBJQpaMLO0KBbnF-dRHEDDQ@mail.gmail.com>	<4E0C3A01.7040006@tagnet.ru>	<BANLkTinNi0=5G00vmRaPZmj-WnVDOAxKYc+6CwsuZgYF_cdigQ@mail.gmail.com>	<4E0C6B9B.8080604@tagnet.ru>
	<BANLkTimMMRH9d2krF+TyAPy=9TkMRT1uNA@mail.gmail.com>
Message-ID: <4E0C8BD3.6090807@tagnet.ru>

Hello!

     Would You please tell me what IOS version have You used?
> Hi Boris,
>
> I had some trouble with 28xx series before, but I was trying to use it
> as a PSTN-2-SIP gateway. Try to see if the firmware is up-to-date, as
> there were some SIP protocol related glithces in the code fixed in
> newer firmware.
>
> Kerem
>
>
> On Thu, Jun 30, 2011 at 3:27 PM, Boris Kovalenko<boris at tagnet.ru>  wrote:
>> Hello!
>>
>>      Have reread your post. So 100 Trying may be (or may be not) followed by
>> 180/183. And in case of 180/183 early media will be processed (one way
>> only?). And full media will be only after the 200 OK from Cisco side. It
>> this right? In other words - Cisco *MUST* respond with 100 Trying and 200 OK
>> at least?
>>
>>
>> Btw, this is the latest SIP spec, better than the draft spec for 183 you
>> posted before.
>>
>> http://www.ietf.org/rfc/rfc3261.txt
>>
>> -Steve
>>
>>
>>
>> On 30 June 2011 09:55, Boris Kovalenko<boris at tagnet.ru>  wrote:
>>> Hello!
>>>
>>> Yes, this is all. INVITE, 100 and RTP. No other SIP messages. So my
>>> knowledge of SIP if right. And customer lies to me or forgot to configure
>>> something :)
>>>
>>> Is that all the signalling?
>>>
>>> 100 Trying only acknowledges the INVITE.
>>>
>>> That *may* be followed by 183 or 180 which'll indicate ringing. The Cisco
>>> would only be able to send early media if that was a 183/SDP or 180/SDP.
>>>
>>> The voice will only start when it sends 200/SDP which is when the call is
>>> answered.
>>>
>>> -Steve
>>>
>>>
>>> On 30 June 2011 06:00, Boris Kovalenko<boris at tagnet.ru>  wrote:
>>>> Hello!
>>>>
>>>>      I have a customers with Cisco 2821 which want to use IVR on it. He
>>>> has configured Cisco and told me there is no voice when number is
>>>> called. The configuration is:
>>>>
>>>> Cisco 2821<-- FreeSwitch (full proxy mode) -->  PSTN
>>>>
>>>> I looked at tcpdump and the call flow is:
>>>> FS (INVITE) -->  Cisco 2821
>>>> Cisco 2821 (100 Trying) -->  FS
>>>> Cisco 2821 (RTP) -->  FS
>>>>
>>>> I think the call flow is wrong in that case and SIP100 can not go
>>>> throught Freeswitch
>>>> (http://jdrosen.net/papers/draft-donovan-mmusic-183-00.txt) and there
>>>> should be SIP180 to get it work. But customers said this is OK and he
>>>> has no troubles with same configuration with other provider. What may I
>>>> do in this sutiation? Any specific Freeswitch configuration?
>>>>
>>>> P.S. FreeSWITCH Version 1.0.head (git-1c95ad9 2011-01-20 22-43-50 -0300)
>>>>
>>>> --
>>>> Regards,
>>>> Boris
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> --
>>> ? ?????????,
>>>    ????? ?????????
>>>    ??? "??????"
>>>    ???. +7 (3435) 230001
>>>    ???? +7 (3435) 230005
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> --
>> Regards,
>> Boris
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>


-- 
Regards,
Boris




From boris at tagnet.ru  Thu Jun 30 18:54:21 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 30 Jun 2011 20:54:21 +0600
Subject: [Freeswitch-users] SIP 100 response and media
In-Reply-To: <BANLkTim4hoBnmSXjRDyqqU4z7YzQ7s1CnQ@mail.gmail.com>
References: <4E0C02FB.2060703@tagnet.ru>	<BANLkTikh_DOBBJQpaMLO0KBbnF-dRHEDDQ@mail.gmail.com>	<4E0C3A01.7040006@tagnet.ru>	<BANLkTinNi0=5G00vmRaPZmj-WnVDOAxKYc+6CwsuZgYF_cdigQ@mail.gmail.com>	<4E0C6542.1040206@tagnet.ru>
	<BANLkTim4hoBnmSXjRDyqqU4z7YzQ7s1CnQ@mail.gmail.com>
Message-ID: <4E0C8E1D.2000506@tagnet.ru>

Hello!

     Thank You, Steve!
> Yes, either that or 200 (you can skip ringing to go straight to being 
> answered).
>
> Generally with 183/180 you'll get ringback but not hear voice in early 
> media, you wouldn't normally hear voice until the 200 OK answers the call.
>
> Media can't start up without SDP in the body of the 180/183/200. SDP 
> contains the IP/port used for media and the selected codec. Media 
> can't work without those details. Sometimes you'll see 180/183 without 
> SDP there'll still be no early media containing ringback, in that case 
> the client'll usually generate its own ringback but you'll always see 
> SDP in the 200 OK.
>
> -Steve
>
>
>
>
> On 30 June 2011 13:00, Boris Kovalenko <boris at tagnet.ru 
> <mailto:boris at tagnet.ru>> wrote:
>
>     Hello!
>
>         But even reading this RFC, I found that 100 Trying should be
>     followed by one of 18x response to get media channels connected,
>     isn't?
>
>>     Btw, this is the latest SIP spec, better than the draft spec for
>>     183 you posted before.
>>
>>     http://www.ietf.org/rfc/rfc3261.txt
>>
>>     -Steve
>>
>>
>>
>>     On 30 June 2011 09:55, Boris Kovalenko <boris at tagnet.ru
>>     <mailto:boris at tagnet.ru>> wrote:
>>
>>         Hello!
>>
>>         Yes, this is all. INVITE, 100 and RTP. No other SIP messages.
>>         So my knowledge of SIP if right. And customer lies to me or
>>         forgot to configure something :)
>>
>>
>>>         Is that all the signalling?
>>>
>>>         100 Trying only acknowledges the INVITE.
>>>
>>>         That *may* be followed by 183 or 180 which'll indicate
>>>         ringing. The Cisco would only be able to send early media if
>>>         that was a 183/SDP or 180/SDP.
>>>
>>>         The voice will only start when it sends 200/SDP which is
>>>         when the call is answered.
>>>
>>>         -Steve
>>>
>>>
>>>         On 30 June 2011 06:00, Boris Kovalenko <boris at tagnet.ru
>>>         <mailto:boris at tagnet.ru>> wrote:
>>>
>>>             Hello!
>>>
>>>                 I have a customers with Cisco 2821 which want to use
>>>             IVR on it. He
>>>             has configured Cisco and told me there is no voice when
>>>             number is
>>>             called. The configuration is:
>>>
>>>             Cisco 2821 <-- FreeSwitch (full proxy mode) --> PSTN
>>>
>>>             I looked at tcpdump and the call flow is:
>>>             FS (INVITE) --> Cisco 2821
>>>             Cisco 2821 (100 Trying) --> FS
>>>             Cisco 2821 (RTP) --> FS
>>>
>>>             I think the call flow is wrong in that case and SIP100
>>>             can not go
>>>             throught Freeswitch
>>>             (http://jdrosen.net/papers/draft-donovan-mmusic-183-00.txt)
>>>             and there
>>>             should be SIP180 to get it work. But customers said this
>>>             is OK and he
>>>             has no troubles with same configuration with other
>>>             provider. What may I
>>>             do in this sutiation? Any specific Freeswitch configuration?
>>>
>>>             P.S. FreeSWITCH Version 1.0.head (git-1c95ad9 2011-01-20
>>>             22-43-50 -0300)
>>>
>>>             --
>>>             Regards,
>>>             Boris
>>>
>>>
>>>
>>>             _______________________________________________
>>>             Join us at ClueCon 2011, Aug 9-11, Chicago
>>>             http://www.cluecon.com 877-7-4ACLUE
>>>
>>>             FreeSWITCH-users mailing list
>>>             FreeSWITCH-users at lists.freeswitch.org
>>>             <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>             http://www.freeswitch.org
>>>
>>>
>>>
>>>         _______________________________________________
>>>         Join us at ClueCon 2011, Aug 9-11, Chicago
>>>         http://www.cluecon.com  877-7-4ACLUE
>>>
>>>         FreeSWITCH-users mailing list
>>>         FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>         http://www.freeswitch.org
>>
>>
>>         -- 
>>         ? ?????????,
>>            ????? ?????????
>>            ??? "??????"
>>            ???. +7 (3435) 230001
>>            ???? +7 (3435) 230005
>>
>>
>>         _______________________________________________
>>         Join us at ClueCon 2011, Aug 9-11, Chicago
>>         http://www.cluecon.com 877-7-4ACLUE
>>
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>     _______________________________________________
>>     Join us at ClueCon 2011, Aug 9-11, Chicago
>>     http://www.cluecon.com  877-7-4ACLUE
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>     -- 
>     Regards,
>     Boris
>
>
>
>     _______________________________________________
>     Join us at ClueCon 2011, Aug 9-11, Chicago
>     http://www.cluecon.com 877-7-4ACLUE
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Regards,
Boris


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From covici at ccs.covici.com  Thu Jun 30 19:06:39 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 30 Jun 2011 11:06:39 -0400
Subject: [Freeswitch-users] segfault after 407 response
In-Reply-To: <4832486C-1E5B-47F0-90BB-0C86CF41C2F3@freeswitch.org>
References: <23273.1309436827@ccs.covici.com>
	<BANLkTi=gMMMaLico_5DdCYzEO875+cAH2g@mail.gmail.com>
	<4524.1309443042@ccs.covici.com>
	<4832486C-1E5B-47F0-90BB-0C86CF41C2F3@freeswitch.org>
Message-ID: <11999.1309446399@ccs.covici.com>

Well, that is interesting, because in the back trace which I did, a lot
of things said they were optimized out.  I did not change the default
compile in any way, but I thought I remembered having to change CCFLAGS
in the past.

Brian West <brian at freeswitch.org> wrote:

> Unless you did something you can dump core with our default with 'ulimit -c unlimited' because we do NOT compile optimized by default.
> 
> /b
> 
> On Jun 30, 2011, at 10:10 AM, covici at ccs.covici.com wrote:
> 
> > I could not get a core dump, but I can run under gdb and get traces, but
> > I have to compile without optimization and with -g gdb, so are the
> > traces of any use without that?
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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

         John Covici
         covici at ccs.covici.com


From wes-fs at 499x.com  Thu Jun 30 19:10:39 2011
From: wes-fs at 499x.com (Wes)
Date: Thu, 30 Jun 2011 10:10:39 -0500
Subject: [Freeswitch-users] significant delay waiting for response after
	playAndGetDigits()
Message-ID: <4E0C91EF.3090200@499x.com>

I am experiencing a delay waiting for the system to respond in a lua 
script that is using playAndGetDigits.  I press a key while the voice is 
still speaking, and it continues to speak for a few seconds before 
moving to the part of the script for the particular keypress.

Also, related: sometimes it speaks the "invalid" message even though the 
key pressed is a valid one, and then immediately after speaking the 
invalid message, it continues with the processing for the key that was 
pressed.  So I know I didn't press an invalid key.

--        session:execute('flush_dtmf');
         session:flushDigits()
         local digits = session:playAndGetDigits(1, 1, 3, 2000, "#", 
"phrase:play_submit_rerecord:1:2:3", invalid, "\\d{1}")
         if ( digits == "1") then

ETC

here is the phrase:

<macro name="play_submit_rerecord">
<input pattern="^([0-9#*]):([0-9#*]):([0-9#*])$">
<match>
<action function="speak-text"
                 data="To listen to your dictation, press $1, To submit 
your dictation , press $2, To discard your dictation and start over, 
press $3."/>
</match>
</input>
</macro>

Is the speak-text just too cpu intensive and causing it to be slow to 
respond to the DTMF?

Any help would be appreciated... thanks!


From wstephen80 at gmail.com  Thu Jun 30 19:21:22 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Thu, 30 Jun 2011 17:21:22 +0200
Subject: [Freeswitch-users] Optimize Freeswitch build
Message-ID: <BANLkTikiD8FgPGk35EVvc5BTJNUceQhscg@mail.gmail.com>

Hi all,
it's possible to optimize the Freeswitch build to maximize the performance?
Or this is the default configuration?
I have not found any reference to this argument.

Stephen
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From curriegrad2004 at gmail.com  Thu Jun 30 19:45:48 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Thu, 30 Jun 2011 08:45:48 -0700
Subject: [Freeswitch-users] Optimize Freeswitch build
In-Reply-To: <BANLkTikiD8FgPGk35EVvc5BTJNUceQhscg@mail.gmail.com>
References: <BANLkTikiD8FgPGk35EVvc5BTJNUceQhscg@mail.gmail.com>
Message-ID: <BANLkTikpZb9SDjF2fz3hmCQKk3DGHe1QMQ@mail.gmail.com>

You can always use the cflags and cxxflags "-g -O2 -march=proc" to
optimize FreeSwitch altogether during the build process.

On Thu, Jun 30, 2011 at 8:21 AM, Stephen Wilde <wstephen80 at gmail.com> wrote:
> Hi all,
> it's possible to optimize the Freeswitch build to maximize the performance?
> Or this is the default configuration?
> I have not found any reference to this argument.
> Stephen
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From wstephen80 at gmail.com  Thu Jun 30 20:11:10 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Thu, 30 Jun 2011 18:11:10 +0200
Subject: [Freeswitch-users] Optimize Freeswitch build
In-Reply-To: <BANLkTikpZb9SDjF2fz3hmCQKk3DGHe1QMQ@mail.gmail.com>
References: <BANLkTikiD8FgPGk35EVvc5BTJNUceQhscg@mail.gmail.com>
	<BANLkTikpZb9SDjF2fz3hmCQKk3DGHe1QMQ@mail.gmail.com>
Message-ID: <BANLkTime0Qq70k3Px3ZcNsg48mOTXyT-8w@mail.gmail.com>

Thank you for the info, but how I can set this flags?
I have to change the Makefile?
I build Freeswitch with a "make install".


On Thu, Jun 30, 2011 at 5:45 PM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:

> You can always use the cflags and cxxflags "-g -O2 -march=proc" to
> optimize FreeSwitch altogether during the build process.
>
> On Thu, Jun 30, 2011 at 8:21 AM, Stephen Wilde <wstephen80 at gmail.com>
> wrote:
> > Hi all,
> > it's possible to optimize the Freeswitch build to maximize the
> performance?
> > Or this is the default configuration?
> > I have not found any reference to this argument.
> > Stephen
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jan.berger at video24.no  Thu Jun 30 20:17:04 2011
From: jan.berger at video24.no (Jan Berger)
Date: Thu, 30 Jun 2011 18:17:04 +0200
Subject: [Freeswitch-users] Optimize Freeswitch build
In-Reply-To: <BANLkTime0Qq70k3Px3ZcNsg48mOTXyT-8w@mail.gmail.com>
References: <BANLkTikiD8FgPGk35EVvc5BTJNUceQhscg@mail.gmail.com>
	<BANLkTikpZb9SDjF2fz3hmCQKk3DGHe1QMQ@mail.gmail.com>
	<BANLkTime0Qq70k3Px3ZcNsg48mOTXyT-8w@mail.gmail.com>
Message-ID: <BD4247AF5F8E43EFA37E14AF6CD74DF0@dell9400>

Assuming "maximum performance" is for the purpose of getting more calls when
it hardly matters unless CPU is your actual bottleneck. 

 

Jan

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stephen
Wilde
Sent: 30. juni 2011 18:11
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Optimize Freeswitch build

 

Thank you for the info, but how I can set this flags?

I have to change the Makefile?

I build Freeswitch with a "make install".

 

On Thu, Jun 30, 2011 at 5:45 PM, curriegrad2004 <curriegrad2004 at gmail.com>
wrote:

You can always use the cflags and cxxflags "-g -O2 -march=proc" to
optimize FreeSwitch altogether during the build process.


On Thu, Jun 30, 2011 at 8:21 AM, Stephen Wilde <wstephen80 at gmail.com> wrote:
> Hi all,
> it's possible to optimize the Freeswitch build to maximize the
performance?
> Or this is the default configuration?
> I have not found any reference to this argument.
> Stephen

> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

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

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

 

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From curriegrad2004 at gmail.com  Thu Jun 30 20:23:22 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Thu, 30 Jun 2011 09:23:22 -0700
Subject: [Freeswitch-users] Optimize Freeswitch build
In-Reply-To: <BD4247AF5F8E43EFA37E14AF6CD74DF0@dell9400>
References: <BANLkTikiD8FgPGk35EVvc5BTJNUceQhscg@mail.gmail.com>
	<BANLkTikpZb9SDjF2fz3hmCQKk3DGHe1QMQ@mail.gmail.com>
	<BANLkTime0Qq70k3Px3ZcNsg48mOTXyT-8w@mail.gmail.com>
	<BD4247AF5F8E43EFA37E14AF6CD74DF0@dell9400>
Message-ID: <BANLkTik3PfW5wTA0vSkuaysqQmtc8ng30Q@mail.gmail.com>

You enable the optimizations by this following command in the configure stage:
"CFLAGS="-g -O2 -march=native" CXXFLAGS="-g -O2 -march=native" ./configure"

after that simply run a make and you should be good.

On Thu, Jun 30, 2011 at 9:17 AM, Jan Berger <jan.berger at video24.no> wrote:
> Assuming "maximum performance" is for the purpose of getting more calls when
> it hardly matters unless CPU is your actual bottleneck.
>
>
>
> Jan
>
>
>
> ________________________________
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stephen
> Wilde
> Sent: 30. juni 2011 18:11
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Optimize Freeswitch build
>
>
>
> Thank you for the info, but how I can set this flags?
>
> I have to change the Makefile?
>
> I build Freeswitch with a "make install".
>
>
>
> On Thu, Jun 30, 2011 at 5:45 PM, curriegrad2004 <curriegrad2004 at gmail.com>
> wrote:
>
> You can always use the cflags and cxxflags "-g -O2 -march=proc" to
> optimize FreeSwitch altogether during the build process.
>
> On Thu, Jun 30, 2011 at 8:21 AM, Stephen Wilde <wstephen80 at gmail.com> wrote:
>> Hi all,
>> it's possible to optimize the Freeswitch build to maximize the
>> performance?
>> Or this is the default configuration?
>> I have not found any reference to this argument.
>> Stephen
>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From wstephen80 at gmail.com  Thu Jun 30 21:24:10 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Thu, 30 Jun 2011 19:24:10 +0200
Subject: [Freeswitch-users] Optimize Freeswitch build
In-Reply-To: <BANLkTik3PfW5wTA0vSkuaysqQmtc8ng30Q@mail.gmail.com>
References: <BANLkTikiD8FgPGk35EVvc5BTJNUceQhscg@mail.gmail.com>
	<BANLkTikpZb9SDjF2fz3hmCQKk3DGHe1QMQ@mail.gmail.com>
	<BANLkTime0Qq70k3Px3ZcNsg48mOTXyT-8w@mail.gmail.com>
	<BD4247AF5F8E43EFA37E14AF6CD74DF0@dell9400>
	<BANLkTik3PfW5wTA0vSkuaysqQmtc8ng30Q@mail.gmail.com>
Message-ID: <BANLkTi=zUS_xvSUXQon3nHJ8Eg6sFjpzpQ@mail.gmail.com>

Thank you curriegrad2004, I'll rebuild Freeswitch with these settings.

Stephen

On Thu, Jun 30, 2011 at 6:23 PM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:

> You enable the optimizations by this following command in the configure
> stage:
> "CFLAGS="-g -O2 -march=native" CXXFLAGS="-g -O2 -march=native" ./configure"
>
> after that simply run a make and you should be good.
>
> On Thu, Jun 30, 2011 at 9:17 AM, Jan Berger <jan.berger at video24.no> wrote:
> > Assuming "maximum performance" is for the purpose of getting more calls
> when
> > it hardly matters unless CPU is your actual bottleneck.
> >
> >
> >
> > Jan
> >
> >
> >
> > ________________________________
> >
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Stephen
> > Wilde
> > Sent: 30. juni 2011 18:11
> > To: FreeSWITCH Users Help
> > Subject: Re: [Freeswitch-users] Optimize Freeswitch build
> >
> >
> >
> > Thank you for the info, but how I can set this flags?
> >
> > I have to change the Makefile?
> >
> > I build Freeswitch with a "make install".
> >
> >
> >
> > On Thu, Jun 30, 2011 at 5:45 PM, curriegrad2004 <
> curriegrad2004 at gmail.com>
> > wrote:
> >
> > You can always use the cflags and cxxflags "-g -O2 -march=proc" to
> > optimize FreeSwitch altogether during the build process.
> >
> > On Thu, Jun 30, 2011 at 8:21 AM, Stephen Wilde <wstephen80 at gmail.com>
> wrote:
> >> Hi all,
> >> it's possible to optimize the Freeswitch build to maximize the
> >> performance?
> >> Or this is the default configuration?
> >> I have not found any reference to this argument.
> >> Stephen
> >
> >> _______________________________________________
> >> Join us at ClueCon 2011, Aug 9-11, Chicago
> >> http://www.cluecon.com 877-7-4ACLUE
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From anthony.minessale at gmail.com  Thu Jun 30 21:32:27 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 30 Jun 2011 12:32:27 -0500
Subject: [Freeswitch-users] segfault after 407 response
In-Reply-To: <11999.1309446399@ccs.covici.com>
References: <23273.1309436827@ccs.covici.com>
	<BANLkTi=gMMMaLico_5DdCYzEO875+cAH2g@mail.gmail.com>
	<4524.1309443042@ccs.covici.com>
	<4832486C-1E5B-47F0-90BB-0C86CF41C2F3@freeswitch.org>
	<11999.1309446399@ccs.covici.com>
Message-ID: <BANLkTinMM=CNoQD++dvdt1e5rYncEeaDww@mail.gmail.com>

there was a jira on this issue, its already fixed in tree as of like 10 min ago.


On Thu, Jun 30, 2011 at 10:06 AM,  <covici at ccs.covici.com> wrote:
> Well, that is interesting, because in the back trace which I did, a lot
> of things said they were optimized out. ?I did not change the default
> compile in any way, but I thought I remembered having to change CCFLAGS
> in the past.
>
> Brian West <brian at freeswitch.org> wrote:
>
>> Unless you did something you can dump core with our default with 'ulimit -c unlimited' because we do NOT compile optimized by default.
>>
>> /b
>>
>> On Jun 30, 2011, at 10:10 AM, covici at ccs.covici.com wrote:
>>
>> > I could not get a core dump, but I can run under gdb and get traces, but
>> > I have to compile without optimization and with -g gdb, so are the
>> > traces of any use without that?
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> --
> Your life is like a penny. ?You're going to lose it. ?The question is:
> How do
> you spend it?
>
> ? ? ? ? John Covici
> ? ? ? ? covici at ccs.covici.com
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From kerem.erciyes at gmail.com  Thu Jun 30 21:43:10 2011
From: kerem.erciyes at gmail.com (Kerem Erciyes)
Date: Thu, 30 Jun 2011 20:43:10 +0300
Subject: [Freeswitch-users] SIP 100 response and media
In-Reply-To: <4E0C8E1D.2000506@tagnet.ru>
References: <4E0C02FB.2060703@tagnet.ru>
	<BANLkTikh_DOBBJQpaMLO0KBbnF-dRHEDDQ@mail.gmail.com>
	<4E0C3A01.7040006@tagnet.ru>
	<BANLkTinNi0=5G00vmRaPZmj-WnVDOAxKYc+6CwsuZgYF_cdigQ@mail.gmail.com>
	<4E0C6542.1040206@tagnet.ru>
	<BANLkTim4hoBnmSXjRDyqqU4z7YzQ7s1CnQ@mail.gmail.com>
	<4E0C8E1D.2000506@tagnet.ru>
Message-ID: <BANLkTimgdUf6mmQk1E=U9fGnLgQ8cdPAug@mail.gmail.com>

Sorry Boris,

As I do not have access to the device anymore I am unable to check the version.
I remember we started out with 12.4 something and pushed to at least 12.11.

Regards,
Kerem


On Thu, Jun 30, 2011 at 5:54 PM, Boris Kovalenko <boris at tagnet.ru> wrote:
> Hello!
>
> ??? Thank You, Steve!
>
> Yes, either that or 200 (you can skip ringing to go straight to being
> answered).
>
> Generally with 183/180 you'll get ringback but not hear voice in early
> media, you wouldn't normally hear voice until the 200 OK answers the call.
>
> Media can't start up without SDP in the body of the 180/183/200. SDP
> contains the IP/port used for media and the selected codec. Media can't work
> without those details. Sometimes you'll see 180/183 without SDP there'll
> still be no early media containing ringback, in that case the client'll
> usually generate its own ringback but you'll always see SDP in the 200 OK.
>
> -Steve
>
>
>
>
> On 30 June 2011 13:00, Boris Kovalenko <boris at tagnet.ru> wrote:
>>
>> Hello!
>>
>> ??? But even reading this RFC, I found that 100 Trying should be followed
>> by one of 18x response to get media channels connected, isn't?
>>
>> Btw, this is the latest SIP spec, better than the draft spec for 183 you
>> posted before.
>>
>> http://www.ietf.org/rfc/rfc3261.txt
>>
>> -Steve
>>
>>
>>
>> On 30 June 2011 09:55, Boris Kovalenko <boris at tagnet.ru> wrote:
>>>
>>> Hello!
>>>
>>> Yes, this is all. INVITE, 100 and RTP. No other SIP messages. So my
>>> knowledge of SIP if right. And customer lies to me or forgot to configure
>>> something :)
>>>
>>> Is that all the signalling?
>>>
>>> 100 Trying only acknowledges the INVITE.
>>>
>>> That *may* be followed by 183 or 180 which'll indicate ringing. The Cisco
>>> would only be able to send early media if that was a 183/SDP or 180/SDP.
>>>
>>> The voice will only start when it sends 200/SDP which is when the call is
>>> answered.
>>>
>>> -Steve
>>>
>>>
>>> On 30 June 2011 06:00, Boris Kovalenko <boris at tagnet.ru> wrote:
>>>>
>>>> Hello!
>>>>
>>>> ? ? I have a customers with Cisco 2821 which want to use IVR on it. He
>>>> has configured Cisco and told me there is no voice when number is
>>>> called. The configuration is:
>>>>
>>>> Cisco 2821 <-- FreeSwitch (full proxy mode) --> PSTN
>>>>
>>>> I looked at tcpdump and the call flow is:
>>>> FS (INVITE) --> Cisco 2821
>>>> Cisco 2821 (100 Trying) --> FS
>>>> Cisco 2821 (RTP) --> FS
>>>>
>>>> I think the call flow is wrong in that case and SIP100 can not go
>>>> throught Freeswitch
>>>> (http://jdrosen.net/papers/draft-donovan-mmusic-183-00.txt) and there
>>>> should be SIP180 to get it work. But customers said this is OK and he
>>>> has no troubles with same configuration with other provider. What may I
>>>> do in this sutiation? Any specific Freeswitch configuration?
>>>>
>>>> P.S. FreeSWITCH Version 1.0.head (git-1c95ad9 2011-01-20 22-43-50 -0300)
>>>>
>>>> --
>>>> Regards,
>>>> Boris
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> --
>>> ? ?????????,
>>>   ????? ?????????
>>>   ??? "??????"
>>>   ???. +7 (3435) 230001
>>>   ???? +7 (3435) 230005
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> --
>> Regards,
>> Boris
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> --
> Regards,
> Boris
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kerem Erciyes - Sistem Danismani
http://keremerciyes.com


From robert.hadley at teotech.com  Thu Jun 30 21:44:39 2011
From: robert.hadley at teotech.com (Robert Hadley)
Date: Thu, 30 Jun 2011 10:44:39 -0700
Subject: [Freeswitch-users] How to send Outbound Caller ID Name out Sangoma
	A101 PRI
Message-ID: <E826C94A3C6BE44198FAF4D66B5AB4D81BDA7E5903@VA3DIAXVS351.RED001.local>

Hi FS Users,

Is there a way to send the outbound caller ID name when using FreeTDM, wanpipe, and a Sangoma A101 PRI?

I am setting the effective_caller_id_name and _number and the number goes out but not the name.
I also tried setting the origination_caller_id_name on the bridge but that didn't work either.
I receive the incoming caller_id_name in a separate Facility message.

Thanks,
Robert
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From spencer at 5ninesolutions.com  Thu Jun 30 22:40:18 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Thu, 30 Jun 2011 11:40:18 -0700
Subject: [Freeswitch-users] t38-passthru sofia param
Message-ID: <724715C2-4970-4594-9346-167EA8B89C0F@5ninesolutions.com>

Hello all,
I have a few questions regarding the t38-passthru sofia param.  One of the ways we are using FreeSWITCH is as a simple B2BUA in our SBCs.  The topology looks like this:

                                             Freeswitch
                                                /\          \/
FS and Asterisk boxen -> OpenSIPS -> PSTN

The Freeswitch instance is setup with inbound late negotiation and bypass media = true and the dialplan consists of a very simple bridge statement that sends a call back to the proxy.  

The problem is that when someone sends a fax and the GW send a t.38 reINVITE, Freeswitch is sending the original SDP (PCMU in this case) back to Leg A and the fax then fails using PCMU.  Is this parameter needed when bypass media is already enabled?

Thanks,
Spencer




From jerry.richards at teotech.com  Thu Jun 30 22:42:17 2011
From: jerry.richards at teotech.com (Jerry Richards)
Date: Thu, 30 Jun 2011 11:42:17 -0700
Subject: [Freeswitch-users] Call Dropped When Answered If Bridged To
 Internal Extension And PSTN Number Via
Message-ID: <2BF7FB90DF25EA4485949F3AF2B9D69624F481C617@VA3DIAXVS351.RED001.local>

Hello,

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

http://pastebin.freeswitch.org/16634

The log shows many errors of the type:

reading on a session with no media!

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

Thanks,
Jerry
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From wes-fs at 499x.com  Thu Jun 30 22:46:39 2011
From: wes-fs at 499x.com (Wes)
Date: Thu, 30 Jun 2011 13:46:39 -0500
Subject: [Freeswitch-users] significant delay waiting for response after
 playAndGetDigits()
In-Reply-To: <4E0C91EF.3090200@499x.com>
References: <4E0C91EF.3090200@499x.com>
Message-ID: <4E0CC48F.6060703@499x.com>

I've narrowed it down a bit.  I think this may be a bug (memory leak maybe?)

It seems that the pagd function is very responsive on the first use, but 
on each repeated call to this function, it takes longer to respond to 
the user's keypress.

The following test script illustrates the problem.  It asks user to 
press a key, then it starts counting up to 10.  The first time, it stops 
right away after you press a key, telling you what key you pressed, the 
second time, it gets as far as the number 3, the third time, it counts 
up to six, and so on, until it completes the count all the way to 10, 
and then it says you've pressed an invalid entry, even though you indeed 
pressed a key at the beginning of the message.

lua script:
http://pastebin.freeswitch.org/16637

phrase xml:
http://pastebin.freeswitch.org/16636

Log:
http://pastebin.freeswitch.org/16635

Could someone try this example on their own system to see if they could 
duplicate the behavior?

Thanks in advance!


On 6/30/2011 10:10 AM, Wes wrote:
> I am experiencing a delay waiting for the system to respond in a lua
> script that is using playAndGetDigits.  I press a key while the voice is
> still speaking, and it continues to speak for a few seconds before
> moving to the part of the script for the particular keypress.
>
> Also, related: sometimes it speaks the "invalid" message even though the
> key pressed is a valid one, and then immediately after speaking the
> invalid message, it continues with the processing for the key that was
> pressed.  So I know I didn't press an invalid key.
>
> --        session:execute('flush_dtmf');
>           session:flushDigits()
>           local digits = session:playAndGetDigits(1, 1, 3, 2000, "#",
> "phrase:play_submit_rerecord:1:2:3", invalid, "\\d{1}")
>           if ( digits == "1") then
>
> ETC
>
> here is the phrase:
>
> <macro name="play_submit_rerecord">
> <input pattern="^([0-9#*]):([0-9#*]):([0-9#*])$">
> <match>
> <action function="speak-text"
>                   data="To listen to your dictation, press $1, To submit
> your dictation , press $2, To discard your dictation and start over,
> press $3."/>
> </match>
> </input>
> </macro>
>
> Is the speak-text just too cpu intensive and causing it to be slow to
> respond to the DTMF?
>
> Any help would be appreciated... thanks!
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From anthony.minessale at gmail.com  Thu Jun 30 22:50:45 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 30 Jun 2011 13:50:45 -0500
Subject: [Freeswitch-users] significant delay waiting for response after
	playAndGetDigits()
In-Reply-To: <4E0CC48F.6060703@499x.com>
References: <4E0C91EF.3090200@499x.com>
	<4E0CC48F.6060703@499x.com>
Message-ID: <BANLkTinfwDbQ+6EvhgW4jftiuZ0oV5YUOw@mail.gmail.com>

have you monitored your dtmf ?
is it 2833?

Try console loglevel debug and look at the message saying when you got the dtmf.


On Thu, Jun 30, 2011 at 1:46 PM, Wes <wes-fs at 499x.com> wrote:
> I've narrowed it down a bit. ?I think this may be a bug (memory leak maybe?)
>
> It seems that the pagd function is very responsive on the first use, but
> on each repeated call to this function, it takes longer to respond to
> the user's keypress.
>
> The following test script illustrates the problem. ?It asks user to
> press a key, then it starts counting up to 10. ?The first time, it stops
> right away after you press a key, telling you what key you pressed, the
> second time, it gets as far as the number 3, the third time, it counts
> up to six, and so on, until it completes the count all the way to 10,
> and then it says you've pressed an invalid entry, even though you indeed
> pressed a key at the beginning of the message.
>
> lua script:
> http://pastebin.freeswitch.org/16637
>
> phrase xml:
> http://pastebin.freeswitch.org/16636
>
> Log:
> http://pastebin.freeswitch.org/16635
>
> Could someone try this example on their own system to see if they could
> duplicate the behavior?
>
> Thanks in advance!
>
>
> On 6/30/2011 10:10 AM, Wes wrote:
>> I am experiencing a delay waiting for the system to respond in a lua
>> script that is using playAndGetDigits. ?I press a key while the voice is
>> still speaking, and it continues to speak for a few seconds before
>> moving to the part of the script for the particular keypress.
>>
>> Also, related: sometimes it speaks the "invalid" message even though the
>> key pressed is a valid one, and then immediately after speaking the
>> invalid message, it continues with the processing for the key that was
>> pressed. ?So I know I didn't press an invalid key.
>>
>> -- ? ? ? ?session:execute('flush_dtmf');
>> ? ? ? ? ? session:flushDigits()
>> ? ? ? ? ? local digits = session:playAndGetDigits(1, 1, 3, 2000, "#",
>> "phrase:play_submit_rerecord:1:2:3", invalid, "\\d{1}")
>> ? ? ? ? ? if ( digits == "1") then
>>
>> ETC
>>
>> here is the phrase:
>>
>> <macro name="play_submit_rerecord">
>> <input pattern="^([0-9#*]):([0-9#*]):([0-9#*])$">
>> <match>
>> <action function="speak-text"
>> ? ? ? ? ? ? ? ? ? data="To listen to your dictation, press $1, To submit
>> your dictation , press $2, To discard your dictation and start over,
>> press $3."/>
>> </match>
>> </input>
>> </macro>
>>
>> Is the speak-text just too cpu intensive and causing it to be slow to
>> respond to the DTMF?
>>
>> Any help would be appreciated... thanks!
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From wes-fs at 499x.com  Thu Jun 30 23:01:04 2011
From: wes-fs at 499x.com (Wes)
Date: Thu, 30 Jun 2011 14:01:04 -0500
Subject: [Freeswitch-users] significant delay waiting for response after
 playAndGetDigits()
In-Reply-To: <BANLkTinfwDbQ+6EvhgW4jftiuZ0oV5YUOw@mail.gmail.com>
References: <4E0C91EF.3090200@499x.com>	<4E0CC48F.6060703@499x.com>
	<BANLkTinfwDbQ+6EvhgW4jftiuZ0oV5YUOw@mail.gmail.com>
Message-ID: <4E0CC7F0.201@499x.com>

I'm in fs_cli, and I think the loglevel is already at debug, since I set 
it to debug and saw the same messages that I have in the pastebin

http://pastebin.freeswitch.org/16635

in line 12, you can see that it received DTMF 1 right after speaking "One"

but then, the second time I called pagd, you can see that in line 39, it got as far as speaking the number "Three"
And, the third time, in line 81, it gets to speaking "Six", before detecting the DTMF.

So, is that what you mean by monitoring the DTMF?  Even though I press the number immediately when the message says "go", it takes longer and longer each time to respond.


On 6/30/2011 1:50 PM, Anthony Minessale wrote:
> have you monitored your dtmf ?
> is it 2833?
>
> Try console loglevel debug and look at the message saying when you got the dtmf.
>
>
> On Thu, Jun 30, 2011 at 1:46 PM, Wes<wes-fs at 499x.com>  wrote:
>> I've narrowed it down a bit.  I think this may be a bug (memory leak maybe?)
>>
>> It seems that the pagd function is very responsive on the first use, but
>> on each repeated call to this function, it takes longer to respond to
>> the user's keypress.
>>
>> The following test script illustrates the problem.  It asks user to
>> press a key, then it starts counting up to 10.  The first time, it stops
>> right away after you press a key, telling you what key you pressed, the
>> second time, it gets as far as the number 3, the third time, it counts
>> up to six, and so on, until it completes the count all the way to 10,
>> and then it says you've pressed an invalid entry, even though you indeed
>> pressed a key at the beginning of the message.
>>
>> lua script:
>> http://pastebin.freeswitch.org/16637
>>
>> phrase xml:
>> http://pastebin.freeswitch.org/16636
>>
>> Log:
>> http://pastebin.freeswitch.org/16635
>>
>> Could someone try this example on their own system to see if they could
>> duplicate the behavior?
>>
>> Thanks in advance!
>>
>>
>> On 6/30/2011 10:10 AM, Wes wrote:
>>> I am experiencing a delay waiting for the system to respond in a lua
>>> script that is using playAndGetDigits.  I press a key while the voice is
>>> still speaking, and it continues to speak for a few seconds before
>>> moving to the part of the script for the particular keypress.
>>>
>>> Also, related: sometimes it speaks the "invalid" message even though the
>>> key pressed is a valid one, and then immediately after speaking the
>>> invalid message, it continues with the processing for the key that was
>>> pressed.  So I know I didn't press an invalid key.
>>>
>>> --        session:execute('flush_dtmf');
>>>            session:flushDigits()
>>>            local digits = session:playAndGetDigits(1, 1, 3, 2000, "#",
>>> "phrase:play_submit_rerecord:1:2:3", invalid, "\\d{1}")
>>>            if ( digits == "1") then
>>>
>>> ETC
>>>
>>> here is the phrase:
>>>
>>> <macro name="play_submit_rerecord">
>>> <input pattern="^([0-9#*]):([0-9#*]):([0-9#*])$">
>>> <match>
>>> <action function="speak-text"
>>>                    data="To listen to your dictation, press $1, To submit
>>> your dictation , press $2, To discard your dictation and start over,
>>> press $3."/>
>>> </match>
>>> </input>
>>> </macro>
>>>
>>> Is the speak-text just too cpu intensive and causing it to be slow to
>>> respond to the DTMF?
>>>
>>> Any help would be appreciated... thanks!
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>


From jmoran at secureachsystems.com  Thu Jun 30 23:09:56 2011
From: jmoran at secureachsystems.com (Jason Moran)
Date: Thu, 30 Jun 2011 15:09:56 -0400
Subject: [Freeswitch-users] UniMRCP Server as daemon won't connect RTSP
References: <361E98F99D3CC3439EED59BC1924ED6950815F@SERVER2003.SecuReachSystems.local>
	<EE20009F-717C-4612-9E6C-612DE77AA8CA@gmail.com>
Message-ID: <361E98F99D3CC3439EED59BC1924ED695081F7@SERVER2003.SecuReachSystems.local>

I have a solution to the unimrcpserver daemon mode! strace helped me narrow it down (also turning on unimrcp?s logger.xml settings to full DEBUG).

Final solution was to execute with:

./unimrcpserver ?r /usr/local/unimrcp/ -d

 

Why? I noticed that unimrcpserver failed whenever I started it from elsewhere when using the full path (/usr/local/unimrcp/bin/unimrcpserver ). It couldn?t find various relative config files. So I downloaded the unimrcp source to see where it was looking  (?../?). It turns out there is a startup switch called ?r which allows you to put the full path to the default starting directory.  

 

It always worked in SSH because I was always starting it from in the bin directory with ./unimrcpserver. The ?../? that it defaults with found the correct directories. However, by starting it in daemon mode it was not able to reach the same directory structure from the same starting point. The fully qualified path made it work.

 

-Jason Moran

 

From: Steven Ayre [mailto:steveayre at gmail.com] 
Sent: Tuesday, June 28, 2011 2:47 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] UniMRCP Server as daemon won't connect RTSP

 

Have you tried doing a strace or gcore of it when it's running with --daemon?

 

If it's running but not listening/responding it could be sonething like a locking problem. Tracing might show you what it is (or isnt) doing. Pid file could be one culprit.

Steve on iPhone

 

 


On 27 Jun 2011, at 22:52, "Jason Moran" <jmoran at secureachsystems.com> wrote:

	I?ve been playing around with IVONA?s new unimrcp plugin with unimrcp 1.0.0 (r1725) on OpenSuse 11.4

	 

	Anyway, when I start it up with the following, it stays in the foreground of my SSH w/ the following command:

	./unimrcpserver

	That works! However, since it?s in the foreground when I close the SSH window, it kills the process.

	 

	If I use the so-called daemon mode (-d or --daemon) it says it is going into daemon mode, but FS will immediately return that it ?Failed to Connect to RTSP Server...? at the IP:port I specified. The unimrcpserver process is running, but doesn?t seem to respond to anything.

	./unimrcpserver ?d

	./unimrcpserver --daemon

	 

	If I attempt to background it by either using ?&? or ctrl-z it says [1]+  Stopped  ./unimrcpserver and FS will successfully make the RTSP connection but then nothing will happen. Mod_unimrcp will spit out a warning about ?MRCP session has not opened after 5000 ms?

	./unimrcpserver &

	Those are hard to kill. But when I kill -9 it then FS will finally remove the MRCP handle and tell me that it couldn?t allocate the speech engine.

	 

	Lastly, I can nohup it, but then unimrcpserver eats up 95%+ of the CPU (instead of 1-3% as it does when I have it in the foreground), which it does not do when it runs in any other modes. It also makes a very, very large nohup.out file that keeps on growing. Even if I tell it to not log.

	nohup ./unimrcpserver &

	nohup ./unimrcpserver ?o 0 &

	 

	Ideas??
	Thanks,

	Jason Moran

	 

	_______________________________________________
	Join us at ClueCon 2011, Aug 9-11, Chicago
	http://www.cluecon.com 877-7-4ACLUE
	
	FreeSWITCH-users mailing list
	FreeSWITCH-users at lists.freeswitch.org
	http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
	UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
	http://www.freeswitch.org

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From avi at avimarcus.net  Thu Jun 30 23:19:39 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 30 Jun 2011 22:19:39 +0300
Subject: [Freeswitch-users] ESL Send MESSAGE_WAITING Error - Cannot find
	profile
In-Reply-To: <BANLkTikHzwA0KGyBXAXxOwNid1DfMZVMKA@mail.gmail.com>
References: <BANLkTimGcbOiDUDgkRxbHZyPYASTidbVjQ@mail.gmail.com>
	<BANLkTikHzwA0KGyBXAXxOwNid1DfMZVMKA@mail.gmail.com>
Message-ID: <BANLkTimYvT-apQ7BZS3FFR8EkpEpht2ohw@mail.gmail.com>

Anthony,

FreeSWITCH Version 1.0.head (git-3e6cca9 2011-06-28 10-27-00 -0700)

It still produces an error:  "sofia_presence.c:432 Cannot find profile
20064.domain.com"

-Avi

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


From spencer at 5ninesolutions.com  Thu Jun 30 23:25:50 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Thu, 30 Jun 2011 12:25:50 -0700
Subject: [Freeswitch-users] t38-passthru sofia param
In-Reply-To: <724715C2-4970-4594-9346-167EA8B89C0F@5ninesolutions.com>
References: <724715C2-4970-4594-9346-167EA8B89C0F@5ninesolutions.com>
Message-ID: <A9E1BFB8-8BE0-4ED1-86C9-E84A3ACC4AD2@5ninesolutions.com>

Just a quick follow up.  The fact that the reINVITE wasn't being passed correctly across the bridge was pilot error.. I had enable-soa=false on the profile.  I am however a little confused as when to use the t38-passthru parameter.    Is it needed even when doing bypass media or proxy media?

On Jun 30, 2011, at 11:40 AM, Spencer Thomason wrote:

> Hello all,
> I have a few questions regarding the t38-passthru sofia param.  One of the ways we are using FreeSWITCH is as a simple B2BUA in our SBCs.  The topology looks like this:
> 
>                                             Freeswitch
>                                                /\          \/
> FS and Asterisk boxen -> OpenSIPS -> PSTN
> 
> The Freeswitch instance is setup with inbound late negotiation and bypass media = true and the dialplan consists of a very simple bridge statement that sends a call back to the proxy.  
> 
> The problem is that when someone sends a fax and the GW send a t.38 reINVITE, Freeswitch is sending the original SDP (PCMU in this case) back to Leg A and the fax then fails using PCMU.  Is this parameter needed when bypass media is already enabled?
> 
> Thanks,
> Spencer
> 
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From sharad2710 at hotmail.com  Thu Jun 30 08:35:57 2011
From: sharad2710 at hotmail.com (sharad garg)
Date: Thu, 30 Jun 2011 10:05:57 +0530
Subject: [Freeswitch-users] (no subject)
Message-ID: <SNT126-W63C9737241DBFF43ACCCCEC6580@phx.gbl>


Hi to all,
 
This is just let to know this forum that we have completed the work on 
Alarm Service. All this is done using our great Freeswitch only.

The GUI also is designed for the same.

You can have a view at the attached PPT for the details.

Thanks to Freeswitch developers & community off course who helped us in this project directly or indirectly.

Regards
Sharad


 		 	   		  
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From rzhang at gosilverplus.com  Thu Jun 30 21:59:31 2011
From: rzhang at gosilverplus.com (ran zhang)
Date: Thu, 30 Jun 2011 10:59:31 -0700
Subject: [Freeswitch-users] please help!!! how to set flag 'endconf' in
	bridging conference
In-Reply-To: <E826C94A3C6BE44198FAF4D66B5AB4D81BDA7E5903@VA3DIAXVS351.RED001.local>
References: <E826C94A3C6BE44198FAF4D66B5AB4D81BDA7E5903@VA3DIAXVS351.RED001.local>
Message-ID: <4E0CB983.8060407@gosilverplus.com>

hi all:

I'm trying to creating a conference, so when first member enters the
conference, he has to invite another members
and have at least 1 other member to join to have the conference established, so i'm
using bridging conference.

I need this conference to be terminated when the original creator of the
conference leaves no matter how many members are still left in the conference.

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

can someone tell me what to do to solve this problem or get around?  the key
is i only want the original member to be able to terminate the conference
when he leaves, not other members assuming there are at least 2 members.



From netcentrica at gmail.com  Thu Jun 30 12:54:51 2011
From: netcentrica at gmail.com (Mateusz Bartczak)
Date: Thu, 30 Jun 2011 10:54:51 +0200
Subject: [Freeswitch-users] Freeswitch dynamic routing of all calls
Message-ID: <BANLkTimuCjRcVccyrPpgQeY6N+UamyNQ=A@mail.gmail.com>

Hi all

I'm new to FS and I would like to know is it possible to implement following
scenario:

1. User dials number
2. Routing script detects outgoing call event. Every call needs to be
handled by routing script.
3. Routing script takes in input: user name, domain, dialed number. Than it
query database to find best SIP trunk to route the call, it also checks
destination price per minute and calculates maximum call time for prepaid
user.
4. Routing script output is: SIP trunk to use, SIP call parameters (ie.
callerid), maximum call duration
5. FS read output from routing script and make call using returned
parameters

Preferred routing implementation technology: background running unix deamon
written in Java or PHP. Connection with FS via socket.

Event routing script will be multi-threaded, must be able to deal with a lot
of calls in parallel and processing of one call should not block processing
of other calls (I have this problem with Yate voip server, and that's really
big problem)

Is it possible to do this using FS?
Any advices where to search for additional info? I know that there is event
handler but can it return "dialstring" for outgoing call events?
Some code examples?

I will really appreciate your help
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From rzhang at gosilverplus.com  Thu Jun 30 23:02:11 2011
From: rzhang at gosilverplus.com (ran zhang)
Date: Thu, 30 Jun 2011 12:02:11 -0700
Subject: [Freeswitch-users] how to send text messages between sip phones via
	freeswitch
In-Reply-To: <4E0CC48F.6060703@499x.com>
References: <4E0C91EF.3090200@499x.com> <4E0CC48F.6060703@499x.com>
Message-ID: <4E0CC833.6050005@gosilverplus.com>

can someone tell me if it is is possible to  send text messages between 
sip phones via freeswitch?


From kris at kriskinc.com  Thu Jun 30 23:59:14 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 30 Jun 2011 15:59:14 -0400
Subject: [Freeswitch-users] how to send text messages between sip phones
 via freeswitch
In-Reply-To: <4E0CC833.6050005@gosilverplus.com>
References: <4E0C91EF.3090200@499x.com> <4E0CC48F.6060703@499x.com>
	<4E0CC833.6050005@gosilverplus.com>
Message-ID: <BANLkTikOXOb6HpYawFuBb8YbcrZCAaPDOg@mail.gmail.com>

FreeSWITCH supports MESSAGE if that's what you're asking.

On Thu, Jun 30, 2011 at 3:02 PM, ran zhang <rzhang at gosilverplus.com> wrote:
> can someone tell me if it is is possible to ?send text messages between
> sip phones via freeswitch?
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner